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[4] CCVP GWGK Quick Reference Sheets by Brion S. Washington
INTRODUCTION
Introduction For as long as anyone alive can remember, there has always been a way to pick up some sort of phone for chats with distant relatives or with companies that offer customer support. The protocols for this have been ever changing since the invention of the telephone. The old days of human operators gave way to central office switches, which in turn saw the uptake of private branch exchange (PBX) use in companies. Telecommunication methods are ever changing to meet customer expectations and to generate new services (and thus money). In the early stages of voice communications, the rise of major telco companies brought numerous changes. For the most part, the changes proved to be beneficial to the growth of the industry. Among these benefits was the financial power to expand services to all parts of the world. Then, no matter where you lived, you probably had access to a phone. This almost universal access allowed providers to generate more revenue and reinvest in emerging technologies and research. The downside to this was that small companies could not compete with the giants of the field. When the smaller companies couldn’t get a foothold into the market, larger companies continued their dominance. This market lockout started a selfish trend: Companies could do what they wanted with their equipment with little regard for client needs or interoperability with other companies. This monopoly created an environment in which the equipment was proprietary. Proprietary equipment, such as a PBX, from one company is probably not going to be interoperable with another company’s PBX. The purchasing company is now tied to one brand of equipment.
Telecommunication monopolies have been declining for years. Everywhere you look, smaller companies are evolving, while creating new services and functionality for voice communications. A safe guess is that a few hundred billion phone calls are completed every year in the world. With the customer pool consisting of hundreds of millions, companies are competing to add services and growing their client base. Because smaller companies can now enter the market and challenge the established companies, proprietary networks of the past will be just that: of the past. History shows that customer demand drives technology and new inventions. One demand has been for the intertwining of voice and data networks. But, most companies must answer to shareholders, and shareholders want a return on investment (ROI). The demand for the intertwining of voice and data networks, combined with the need to cut costs (and thus increase ROI), has helped spark the Voice over IP (VoIP) revolution. Modern telecommunications use a variety of protocols and devices. VoIP was one of the technologies created to help combine voice and data networks. VoIP is a method of sending voice and video over data networks. VoIP has emerged as the de facto way for companies to send calls. Within VoIP, there are a few protocols to choose from, as follows: n
H.323
n
SIP (Session Initiation Protocol)
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[5] CCVP GWGK Quick Reference Sheets by Brion S. Washington
PURPOSE OF THIS GUIDE
n
MGCP (Media Gateway Control Protocol)
n
SGCP (Simple Gateway Control Protocol)
n
H.248/MEGACO
Each of these protocols has its advantages and disadvantages. The protocols have different components to provide services. The components can be grouped by the function they serve. Examples of these components include gatekeepers (GK), gateways (GW), endpoints, circuits, call control, and different signaling types. Those who want to implement some type of VoIP need to understand which protocol best suits their voice needs.
Purpose of This Guide This guide is intended to help readers pass the Gateway Gatekeeper (GWGK) exam required for the Cisco Certified Voice Professional (CCVP) and other voice-related certifications. The CCVP certification is valid for three years. After passing the CCVP, you may continue on to Cisco Certified Internetwork Export (CCIE) Voice certification. I hope readers find this guide useful as a preparation tool if they choose to pursue the CCIE Voice certification.
As a tester, you must be able to decide which functions to configure based on the requirements. It is not this guide’s intention to try to explain or even mention all the possible options that can be configured with Cisco GKs and GWs. Instead, I try to give you the information you must have, presented in such a way that you gain a full understanding of the subject matter.
Who Should Read This Guide? Anyone who wants to augment his or her study material for the GWGK exam is the target audience for this guide. However, my intention also is to provide a deeper level of understanding for those who want to read a concise document to gain a fundamental understanding of GW and GK operation in Cisco environments. This guide does not delve into GK/GW minutiae. Instead, it provides a solid foundation on which you can build. For those readers who plan to prepare for additional certifications, this short cut could supplement your study material for other voice-related certifications (including CCIE Voice). —Brion S. Washington
In this guide, you will find the necessary information to prepare for your exam, as outlined at http://www.cisco.com. The 642-452 Implementing Cisco Voice Gateways and Gatekeepers exam tests your ability to configure and set up various functions on these Cisco devices.
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GATEKEEPERS
Gatekeepers In the VoIP world of H.323, the GK plays a vital role if included in the network. The use of a GK is optional; if chosen, the GK must perform certain functions. Gatekeepers act like the brains in an H.323 network. They provide network access, bandwidth management, address translation, accounting, and dial plans. Gatekeepers allow the simple configurations of the majority of devices on your network, while requiring only a few devices to hold the majority of configurations. This architecture keeps call routing, security, and administration centralized to a few devices. Because the network only has a handful of devices that need to be configured when changes occur, the likelihood of errors is diminished. NOTE I use VoIP in this short cut to simplify the concept. Gatekeepers will work on any VoXX network using the H.323 protocol suite. Gatekeepers do not concern themselves with the medium or underlying technologies. Gatekeepers only need to make path selections and know whether resources are available for allocation.
When GKs are used in your H.323 network, GKs must/may provide certain mandatory and optional functions. Some of the functions are automatically configured when the GK service is started; others must be configured manually.
Mandatory and Optional Gatekeeper Functions Mandatory Functions Admission control: Controls endpoint admission into the H.323 network using H.225 Registration, Admission, and Status (RAS) messages, as follows: n
Admission Request (ARQ)
n
Admission Confirm (ACF)
n
Admission Reject (ARJ)
Admission control is automatically started when the GK is started. Address translation: Translation of IP addresses to an E.164 phone number and vice versa.
Gatekeeper Communications The GK operates on three different ports depending on whom it is communicating with. TCP port 1718 is used when communicating with other GKs. TCP port 1719 is used for H.323 devices to register with GK in their zones. The next port used is TCP port 1720 for call control.
Example 1: 44.44.1.31 = 402-555-5555 Example 2: 405-999-9978 = 192.168.1.109. Address translation must be configured manually.
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GATEKEEPERS
Bandwidth control: Provides endpoint bandwidth management requirements using H.225 RAS messages. When an endpoint decides it needs a certain amount of bandwidth for the call, it sends a BRQ message to the GK asking for an amount of bandwidth to be set aside for the call. The GK responds with a BCF or BRJ message. n
Bandwidth Request (BRQ)
n
Bandwidth Confirm (BCF)
n
Bandwidth Reject (BRJ)
Optional Functions Call authorization: The GK can restrict access for endpoints or GWs. Bandwidth management: Reject admission when the required bandwidth is not available. The GK can reject calls if there is not enough bandwidth to support the call. All active calls are recorded by the GK to manage the bandwidth within the zone. The GK use the following formula to decide whether it can accept or reject the call request. Available bandwidth = (Total allocated bandwidth) – (Bandwidth used locally) – (Bandwidth used by all alternates)
NOTE Messages appear in the order in which they will happen on a GK-controlled network.
Zone management: Manages all registered endpoints in the zone. This provides centralized control for many devices within the zone. Zone management is automatically started when the GK is started.
Call authorization and bandwidth management must be configured manually. Call management: The GK maintains active call information. Used for resource allocation / route selection. Call management is automatically started. Use show commands to display this information. Call control signaling: The GK may route call signaling messages between endpoints using Gatekeeper Routed Call Signaling (GKRCS). Normally, the GK uses H.225 call signaling. The endpoints send messages directly to other endpoints without using GK signaling. Cisco gatekeepers do not support GKRCS.
© 2007 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 63 for more details.
CCVP GWGK Quick Reference Sheets CCVP GWGK Quick Reference Sheets By Brion S. Washington ISBN: 1587053241 Prepared for Duc Le, Safari ID:
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H.323 CALL SETUP AND CONTROL
H.323 Call Setup and Control Now that you understand more about GKs, let’s cover in more depth how H.323 communicates.
One way to memorize steps of H.323 call setup using a GK: 1. You get RAS-ed by a 225-lb. bully (H.225), which you have no
control over (UDP) (port 1719). A. RAS messages are passed through the GK to each device.
Intrazone Call
B. If no GK is present, H.225 messages are sent between endpoints.
Gatekeeper Infantry
2. You call 931 (Q.931) (H.225) to set up a call with your friends.
V DP
S
22
5
You do not control them (UDP). H.
)
22
5
(U
RA
3. Your 245-lb. friend (H.245) takes control of everyone on the call
RA
S
(TCP) (port 1720). Negotiations take place.
(U
DP
H.
)
4. Your friends (RTP) over on the negotiated streets and talk to the H.225 (Q.931) Call Setup (UDP)
Gateway Bragg
V
H.245 Call Control (TCP)
bully; you have no control over what happens next (UDP). Gateway V Benning
RTP (UDP)
IP
FIGURE 1.1
Call Control and Setup Using a Gatekeeper
© 2007 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 63 for more details.
CCVP GWGK Quick Reference Sheets CCVP GWGK Quick Reference Sheets By Brion S. Washington ISBN: 1587053241 Prepared for Duc Le, Safari ID:
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[9] CCVP GWGK Quick Reference Sheets by Brion S. Washington
H.323 CALL SETUP AND CONTROL
Steps 2 through 4 are direct between endpoints with no GK intervention. Gatekeeper Routed Call Signaling Gatekeeper Infantry
.2 2 H 5 (Q .2 . 45 93 C 1) al C l C al on l S tro etu l( p TC (U P) DP )
P)
AS
25
(U
R
.2
H
H
Gateway Bragg
V
RTP (UDP)
Devices that want to participate on a network must first locate and then register with a GK. There are two ways to discover a GK. During the registration process, it is actually the GWs that register with the GK. All devices on an H.323 network are endpoints. Regular endpoints, such as phones, faxes, and so on, usually do not register directly to GKs. Their requests are forwarded by their GWs. Unicast: The devices use UDP port 1718. Devices are preconfigured with the IP of the GK and can start with the registration process immediately. The GW sends a GRQ to the GK. A GK responds with a gatekeeper confirm (GCF) or gatekeeper rejection (GRJ) message.
P) D P) (U D (U ) AS R up P et (TC 25 .2 lS l H al tro ) C on 31 l C .9 al (Q 5 C 25 4 .2 .2 H H
D
Gatekeeper Discovery Process
Gateway V Benning
Multicast: The devices try to find a GK with the multicast IP 224.0.1.141. The GK responds with GCF or no response at all. If a GW finds that a GK has gone down, it rejects new calls and attempts to find a GK; active (established) calls remain operational.
Registration Process
IP
1. The endpoint sends a registration request to the GK (port 1718
UDP). FIGURE 1.2
Example: GKRCS
2. The GK responds with a registration confirm or reject (RCF or
Now that you know the communication process and the protocol used, let’s discuss how to locate and gain access to the network.
RRJ).
© 2007 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 63 for more details.
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H.323 CALL SETUP AND CONTROL
Unregistration Process
Types of Gatekeepers
1. The endpoint starts the process by sending an unregister request
(URQ). 2. The GK responds with an unregister confirm (UCF) or unregister
reject (URJ).
To provide for the different requirements and functionalities needed, GKs are categorized into different types. Each type allows the GK to handle certain mission-critical tasks. The GK types are active, backup/alternate, and directory, as follows: n
Active GK—Controls all H.323 devices within a zone. This is the first GK used on the network. All messages go through this device; they are then passed on to other GKs as needed.
n
Alternate/backup GK—The second device that handles control of the zone in the event of the failure of the active GK. See the “Redundancy and Availability” section.
n
Directory GK—Hierarchical GK used to promote scalable growth of the network. The directory GK holds information on all zones. This device helps with Location Request (LRQ) messages. Use of a directory GK helps simplify the process of adding or updating GKs, because the directory GK knows the location of all zones and GKs.
Gatekeeper Unregisters Device 1. The GK sends a URQ. 2. The endpoint responds with a UCF or URJ.
The GK recognizes endpoints by the following: n
Email IDs
n
E.164 addresses (telephone numbers)
n
H.323 IDs (user-defined, case sensitive)
If you are having registration issues, make sure you have correctly configured your device with one of them. It is recommended that each endpoint have at least one fully qualified domain name (FQDN) for interzone communication.
When deciding where to put your GKs, you can save yourself a few headaches if you remember a few simple rules: n
Only one active GK per zone.
n
Redundancy (backup GK) should always be used in case of failure.
n
Use a directory GK when your network is big or has potential for large growth.
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CCVP GWGK Quick Reference Sheets CCVP GWGK Quick Reference Sheets By Brion S. Washington ISBN: 1587053241 Prepared for Duc Le, Safari ID:
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H.323 CALL SETUP AND CONTROL
Gatekeeper Topology The GK has a topology that separates areas called zones. Zones are divided into local and remote zones. Local zones are the ones that you have administrative control over or want to classify as local. Cisco GKs support up to 100 local zones, but only 1 local zone is mandatory. Remote zones are zones that you have no administrative control over, or they could be zones you have picked to be remote for various other reasons. The use of remote zones is optional (whereas local zones are not). Gatekeepers must be aware of their capabilities and where to send calls they cannot complete. This information is gathered by use of prefixes. The two types of prefixes are zone prefix and technology prefix. Zone prefixes state which numbers or area codes the device can complete calls to. The prefix could be the area code for Southern California (714), or it could be an extension in the payroll department (3367). You can set up your GK to handle or complete any combination of numbers. When planning your dial plan, you must consider many aspects, from devices used to path selections.
The basic commands needed to set up a functional intrazone GK are not complicated. We will go over a basic GK, and then move on to more complex configurations.
Basic Intrazone Gatekeeper Configuration The steps involved in a basic configuration are as follows: 1. Set up the GK service: Gk-Infantry # configure terminal Gk-Infantry (config) # gatekeeper Gk-Infantry (config-gk) #
2. Set up the GK local zone: Gk-Infantry (config-gk) # zone local GK-Infantry my.army.com
Syntax: zone local gatekeeper-name domain-name 3. Start the GK: Gk-Infantry (config-gk) # no shutdown
Gk_Infantry running-configuration
Technology prefixes are another way to specify what the GK can and cannot do. This type of prefix allows devices to be grouped together by types or services. Examples of types are location, services, and functions. Cisco GKs use technology prefixes to route calls when there is no match for E.164 addresses to a specific GW.
Gk-Infantry# sh run Building configuration... Current configuration : 54281 bytes ! version 12.3 hostname Gk-Infantry ! boot system flash:c7200-js-mz.123-4.T7.bin Gatekeeper zone local Gk-Infantry my.army.com
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H.323 CALL SETUP AND CONTROL
!——Nonrelevant text has been omitted
The zone remote command specifies the remote zone, IP address, and port to use:
After you have a GW or two configured, calls can then take place across your network.
zone remote other-gatekeeper-name other-domain-name other-gatekeeper-ip-address [port-number][cost cost-value [priority priority-value]] [foreign-domain]
No shutdown
!!— Still under gatekeeper configuration
Interzone Gatekeeper Configurations When you need to make calls outside of your zone, your GW sends your request to your local GK. If the local GK does not know where the remote endpoint is, it forwards the request to another GK or queries the directory GK. The zone remote command is used to configure the remote zone.
You can use the following options with zone remote: n
Port-number—Tells what port to use for RAS. Can use between 1 and 65535.
n
Cost—Assigns cost to the zone. Can use between 1 and 100; default is 50.
n
Priority—Sets the priority of the zone. Value is 1 to 100; default is 50.
n
Foreign-domain—Sets the device to a different administrative domain.
Gatekeeper Infantry Gk-Infantry # configure terminal Gk-Infantry (config) # gatekeeper Gk-Infantry (config-gk) # zone local Gk-Infantry my.army.com Gk-Infantry (config-gk) #zone remote Gk-Calvary my.army.com 192.168.2.1 1719 Gk-Infantry (config-gk) # no shutdown
Gatekeeper Calvary Gk-Calvary # configure terminal Gk-Calvary (config) # gatekeeper Gk-Calvary (config-gk) # zone local Gk-Calvary my.army.com Gk-Calvary (config-gk) # zone remote Gk-Infantry my.army.com 192.168.2.3 1719 Gk-Calvary (config-gk) # no shutdown
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H.323 CALL SETUP AND CONTROL
Directory Gatekeepers
Directory Gatekeeper Call Setup
A Cisco implementation is a device called a directory gatekeeper. The directory GK allows for large VoIP networks without having to maintain full meshes between the GKs. When a GK lacks the information to find a remote endpoint, it sends the LRQ to the directory, a lookup occurs, and the directory forwards the LRQ to the correct GK.
The steps involved in completing a directory GK-assisted call are close to interzone, with just a few changes:
3. GK Infantry performs lookup and does not see User B registered.
Directory Gatekeeper Call Setup
GK Infantry performs prefix lookup and finds a wildcard match on Dir-GK; GK Infantry sends LRQ to GK-DIR and a RIP (Request In Progress) to GW-1.
GK-DIR V 3
4
Gatekeeper Infantry
4. GK-Dir performs lookup and finds a match for GK Calvary; sends Gatekeeper Cavalry
5
V 2
9
V
8
Directory Gatekeeper Configuration GW-2
V
11 10
1
User A Ext 4488
Zone 1
FIGURE 1.3
User B Ext 5577
Zone 2
Directory Gatekeeper
an LRQ to GK Calvary. The remaining steps are steps 4-10 in interzone scenario.
V
3 6
7
GW-1
1. User A dials 5577. 2. GW-1 sends GK Infantry ARQ for permission to make the call.
Directory GKs help create a hierarchical network, which helps promote controllable growth in the network. This type of design negates the need for full-mesh configurations between all the GKs. The differences between implementing a directory GK instead of intrazone or interzone are minor. The main thing that differs is the number of commands configured on each device. The GKs are only set up knowing about their prefixes and where to send unknown prefix requests. On the other side, the directory GK knows and is configured with the prefixes for every GK under its influence.
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CCVP GWGK Quick Reference Sheets CCVP GWGK Quick Reference Sheets By Brion S. Washington ISBN: 1587053241 Prepared for Duc Le, Safari ID:
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H.323 CALL SETUP AND CONTROL
Example: Directory Gatekeeper Configuration
Gatekeeper Infantry Configuration
zone local Dir_GK myzone 192.168.2.101
zone local Gk_Infantry mycoolzone 192.168.3.202
zone remote Gk_Calvary mycoolzone 192.168.3.201 1719
zone remote Dir_GK myzone 192.168.2.101 1719 foreign-domain
zone remote Gk_Infantry mycoolzone 192.168.3.202 1719
zone prefix Gk-Infantry 714….
zone prefix Gk-Infantry 714….
zone prefix Gk-Infantry 661….
zone prefix Gk-Infantry 661….
zone prefix Gk-Infantry 310…
zone prefix Gk-Infantry 310….
zone prefix Dir_GK *
zone prefix Gk-Calvary 212…
lrq forward-queries
zone prefix Gk-Calvary 202…
no shutdown
zone prefix Gk-Calvary 312… lrq forward-queries
In the preceding example, calls starting with 714, 661, and 310 are routed through the Infantry GK; calls starting with 212 and 202 are routed through the Calvary GK.
no shutdown
Gatekeeper Calvary Configuration zone local Gk_Calvary mycoolzone 192.168.3.201 zone remote Dir_GK myzone 192.168.2.101 1719 foreign-domain zone prefix Gk-Calvary 212… zone prefix Gk-Calvary 202… zone prefix Gk-Calvary 312… zone prefix Dir_GK * lrq forward-queries no shutdown
If a call starting with 714 comes into the Calvary GK, the GK sees that no zone prefix is configured specifically for 714. The GK then uses the zone prefix wildcard * and routes the request to the directory GK. We can see that the directory GK does have a zone prefix configured for 714 through GK Infantry. The call is then sent to the Infantry GK. When no E.164 addresses are registered, the GK has two routing options: n
Route using the technology prefix in the called number.
n
The GK assigns default GWs for nonregistered addresses and uses default GWs matching the technology prefix assigned.
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H.323 CALL SETUP AND CONTROL
Intrazone
Gatekeeper Placement Now that we have found the GK and understand the call setup process, we’ll look at how and to whom the GK will be providing its services. H.323 devices that are within the control of a GK comprise a zone. The zone is not concerned with the network topology; it covers all devices associated with the GK regardless of location or network segmentation. There is only one active GK per zone. More than one GK can be in a zone, but the rest are in backup mode. Gatekeepers will operate in an intrazone or interzone network. An intrazone network is all devices that are under the GK’s control. All the devices in the zone use the services provided by the GK. Interzone communications are communications between two or more zones. This communication process of GKs can be separated into logical and physical. Logical communication can be one GK controlling multiple zones on the same device. Physical GK communication can involve two or more GKs, physically separated. For those of you from the data world, think of it as multiple Open Shortest Path First (OSPF) processes running on a single router.
Gatekeeper Infantry V V
V
V
V
V
V
V
V
Zone 1
FIGURE 1.4
Intrazone Topology Example, with Both Callers in the Same
Zone
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H.323 CALL SETUP AND CONTROL
Interzone
Locating Endpoints
Gatekeeper Infantry
Gatekeeper Cavalry
The process of locating endpoints not native to your zone is a fairly straightforward process:
V
V
1. Endpoint 1 sends an ARQ to its GK-A, trying to call another
endpoint. V
2. GK-A cannot locate the endpoint and sends an LRQ to GK-B.
V
3. GK-B responds with a location confirm (LCF) message. The LCF V
V
V
V
message includes the IP of the remote GW responsible for the remote endpoint. 4. GK-A responds with an ACF to endpoint 1.
FIGURE 1.5
V
V
Zone 1
Zone 2
Interzone Topology Example, with Two Callers in Different
Zones
The difference between intrazone and interzone is the number of GKs the call must go through and the paths GK messages take. This adds to the complexity of your network; however, this cannot be helped if you are calling outside your zone or administrative authority.
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H.323 CALL SETUP AND CONTROL
Interzone Call Setup
1. User A dials 5577.
The steps involved to complete an interzone call are similar in some ways to an intrazone call, with a few more steps required to complete the call.
2. GW-1 sends an ARQ to GK Infantry for permission to make the
A prefix lookup is done, and GK Calvary is selected. GK Infantry sends an LRQ to GK Calvary and a RIP (Request In Progress) to GW-1.
Interzone Call Setup
Gatekeeper Cavalry
Gatekeeper Infantry 3
V 2
3
8
5. GK Infantry returns with an ACF with the IP address of GW-2. 7
6. GW-1 starts a Q.931 call setup to GW-2 User B’s extension.
6
GW-1
10
V
4. GK Calvary finds User B registered and sends an LCF with
GW-2’s IP address. V
4
5
call. 3. GK Infantry performs lookup and does not see User B registered.
V
GW-2
7. GW-2 asks permission with an ARQ to accept the call from GW-1. 8. GK Calvary responds with an ACF with GW-1’s IP address.
11 9
9. Plain old telephone service (POTS) call setup.
1
10. Q.931 connect. User B Ext 5577
User A Ext 4488
Zone 1
FIGURE 1.6
Zone 2
Interzone Call Steps
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H.323 CALL SETUP AND CONTROL
Advanced Gatekeeper Configuration Options VoIP networks need to be able to adapt to the constantly changing demands of the telecommunication world. The needs of your network might include better resource management, finely-tuned dial plans, or securing resources. The world in which we live today has caused networks to be designed more securely. VoIP security is beyond the scope of this guide, but I do write about ways to tighten resources; which, in turn, can help secure your network a little more. Some people think that to provide security for a network that a massive security plan is needed. This is not necessarily the case in VoIP. Securing a network is a vague term. I will say that securing a network is a multistage plan that should be implemented as simply as possible. A few commands will further your ability to work with a Cisco GK, while serving a dual purpose of optimization and securing your network. The best way to make sure that your calls are going across as smoothly as possible is to limit unwanted (unknown) endpoints and messages from traversing your network. The following commands provide resource-saving techniques and help secure your devices at a basic level. n
Arq reject-unknown-prefix—Forces GKs to reject unknown prefixes from zones not configured.
n
Lrq reject-unknown-prefix—Forces GKs to reject LRQs from nonconfigured zones.
n
Security acl answerarq—Sets up tokenless call authorization for devices outside the GK’s zone. This command calls on an access control list (ACL) from matching, allowed IPs.
n
Zone subnet—Specifies that discovery and registration messages are only accepted from endpoints on the configured subnets. All other nonconfigured subnets are rejected. Syntax: zone subnet local-gatekeeper-name {default | subnetaddress {/bits-in-mask | mask-address}} enable
Example: Security Configuration Access-list 55 permit ip 99.99.99.1 Access-list 55 remark *** Taylor’s
VoIP Network***
Access-list 55 permit ip 88.88.88.1 Access-list 55 remark *** Brittney’s VoIP Network*** Gatekeeper Security acl answerarq 55 No zone subnet Dir_GK myzone.com default enable Zone subnet Dir_GK myzone.com 192.168.2.0
/24 enable
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GATEKEEPER TRANSACTION MESSAGES PROTOCOL (GKTMP)
Gatekeeper Transaction Messages Protocol (GKTMP) GKTMP is used for communications between GKs and external applications where added functions or services are required beyond the capabilities of the GK. GKTMP passes information between Cisco IOS GKs and the external application using ASCII response/request messages based on RAS by way of an application programming interface (API) or over TCP connections. This works in a straightforward manner. Some of the incoming requests for the GK are sent to the external application to fulfill the request. The GK has triggers configured for each external application it will use. These triggers are based on RAS tags and values. Incoming RAS messages are checked against the triggers. If the messages match a trigger, the GK then repackages the message and forwards it to the correct external application. When the external application is finished processing, the results are sent to the GK for further processing, and messages are sent to the requesting endpoint. The main point to remember here is that GKTMP and API are both used to provide this functionality. Your triggers can be manually configured or dynamically configured through API. Static triggers are configured via the command line, whereas dynamic triggers are configured via an external application.
CCVP GWGK Quick Reference Sheets by Brion S. Washington
GKTMP Messages All GKTMP messages are either requests or responses. The GK creates the request, and the external application sends the response. Example: Registration Request message line message header 1 message header 2 message header 3 message body line 1 message body line 2 message body line 3 message body line 4
n
The message line states what kind of request is being sent.
n
The message header is a few lines that contain the external application name, GK name, and version ID. Format is field:value.
n
Blank space is probably used to help with problem resolution.
n
The message body contains the triggers. The message body is optional. Format is tag=value.
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GATEKEEPER TRANSACTION MESSAGES PROTOCOL (GKTMP)
GKTMP Messages ACF (Admission Confirm) ARJ (Admission Reject) ARQ (Admission Request)
CCVP GWGK Quick Reference Sheets by Brion S. Washington
Some messages are only sent as a response from the external application; they are ACF, ARJ, BCF, BRJ, RCF, and RRJ. The GK sends the following messages as requests: DRQ, RAI, and UQ. Also, the application server can use a command URQ to send untriggered, unsolicited URQs to unregister an endpoint from a GK.
BCF (Bandwidth Confirm) BRJ (Bandwidth Reject)
Redundancy and Availability
BRQ (Bandwidth Request)
Now that we have covered the basic and advanced GK functions, let’s turn to ways to keep your network resources available for users. Most networks contain some kind of redundancy and availability to cope with network or device failures. Gatekeepers are not immune to hardware or software failures, so alternative methods must be engineered. Here are a few ways to help with this task.
DRQ (Disengage Request) IRR (Information Request) LCF (Location Confirm) LRJ (Location Reject) LRQ (Location Request) RIP (Request In Progress) RAI (Resource Availability Indication) RCF (Registration Confirm) RRJ (Registration Reject) RRQ (Registration Request) URQ (Unregistration Request)
Gatekeeper Clustering Gatekeeper clustering provides redundancy for zones in the event of a GK failure. This addition to H.323 allows multiple GKs to control a single zone. When the GK starts, each cluster member receives a GRQ from all the other members. The GRQ contains the alternate GK information. The GK opens TCP connections to all cluster members. Clusters can have up to five GKs per clusters. The clusters communicate with Gatekeeper Update Protocol (GUP). When a device registers with its GK, it is provided with two alternate GKs that can take over when the primary GK fails.
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GATEKEEPER TRANSACTION MESSAGES PROTOCOL (GKTMP)
Gatekeeper Update Protocol GUP is used to pass information between GKs in a cluster. The information included is available and used bandwidth, CPU utilization, GK memory, and the number of endpoints registered. The GK’s CPU utilization and memory are used for resource management. If the GK is overloaded, it asks the endpoint to use another GK with the lowest resources used. One of the important things to remember about clustering is that all GKs share registered endpoint information. Therefore, all GKs within the zone know about all endpoints within the cluster, which helps with LRQs.
GUP Messages
n
n
RegistrationIndication—Sent when a connection is made with a new alternate GK, or when a new endpoint registers.
n
UnregistrationIndication—Sent when a GK is down or an endpoint is aged out, or when an endpoint unregisters.
n
ResourceIndication—Sent when the GW sends the GK an RAI message.
Local Zone Clustering zone cluster local cluster-name local-zone-name
Remote Zone Clustering
GUP uses messages to help exchange control and management information within the cluster. Here are the messages and when they are used: n
CCVP GWGK Quick Reference Sheets by Brion S. Washington
AnnouncementIndication—Sent every 30 seconds by default, by all cluster members. The GK updates information on call capacity, CPU load, endpoint load, memory usage, number of active calls, and number of registered endpoints received from the sending GK.
zone cluster remote cluster name [cost cost-value [priority priority-value]] [foreign-domain][invia inbound-gatekeeper] | [outvia outbound gatekeeper]
Specify devices to participate in with local or remote zone element gatekeeper-name ip-address [port].
AnnouncementReject—Sent if a configuration mismatch exists. The GK will terminate the GUP connection with the sender.
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HOT STANDBY ROUTER PROTOCOL
Hot Standby Router Protocol The second method you might think of implementing is Hot Standby Router Protocol (HSRP). For those of you from the data arena, you are probably already overly familiar with HSRP. HSRP allows devices on your network to reach a secondary GW, router, or GK if the primary cannot be communicated with. HSRP is invisible to the end user; this is achieved by setting up a virtual IP address instead of using the real IP of the device. In the voice world, it can be your GK or GW; in the data world, it is your default GW. You must know a few important things when implementing HSRP on your voice network. The main point is that no state information is maintained when failure occurs. Cisco recommends using clustering with GKs, which maintains state information; HSRP is recommend for GWs and directory GKs.
ip address 75.0.0.1 255.0.0.0 standby 2 ip 75.0.0.3 IP**
**enables HSRP, creates group 2, virtual
No standby 2 preempt
**Not recommended for GKs**
standby 2 priority 110 default is 100**
**highest priority will become active,
standby 2 authentication pepsi string, must be identical**
**unencrypted authentication
standby 2 timers 5 15 ! ! **Infantry config** Hostname Infantry-GK Interface Ethernet 0 ip address 75.0.0.2 255.0.0.0
HSRP has different levels of complexity and protection. We will look at the minimum commands needed to enable HSRP. Basic HSRP has a virtual IP address, a virtual MAC address, and a minimum of two devices that belong to this newly created group. The primary GK is in active mode (accepts new requests), and the secondary GK is in standby mode (waiting for the active GK to fail). HSRP uses UDP port 1985. Hello packets are multicast (224.0.0.2), and Cisco routers automatically select the virtual MAC address. Example: Basic Two-GK HSRP Configuration **Calvary config** hostname Calvary-GK interface Ethernet 0
Standby 2 ip 75.0.0.3 Standby preempt Standby authentication pepsi Standby 2 timers 5 15 ! !
The devices on your network would be configured to use 75.0.0.3 as their GK IP. This will provide backup in the event of device failure. You may skip adding the authentication command; this was added to close any security holes on the GK.
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HOT STANDBY ROUTER PROTOCOL
Note
n
This example is a simple HSRP configuration. If you need a more complex configuration or more information, refer to http://www.cisco.com.
show gatekeeper zone prefix Shows which E.164 prefixes are routed to which zones.
n
show gatekeeper zone status Shows zone and configuration values.
n
show gatekeeper cluster Shows which cluster are configured.
n
show gatekeeper performance statistics Shows information on accepted and rejected calls. Will give information based on GK RAS messages. Associated command is clear h323 gatekeeper statistics.
n
clear h323 gatekeeper statistics Resets statistics period.
Verifying Gatekeeper Functions After going through the lengthy process of setting up your network, it is good practice to make sure everything is configured correctly. Verification can provide some quality control and assist with resolving issues. You can use a number of Cisco IOS commands to check your configurations for typos, errors, or mismatched values. The key to any verification process is to know which commands to use and then look at the output slowly line by line. Here are some of the main commands to use first: n
show gatekeeper calls Shows active calls GK is aware of.
n
show gatekeeper endpoints Shows which GWs are registered with the GK.
n
show gatekeeper gw-type-prefix Shows GWs that have registered with a tech prefix and how the GK treats the tech prefixes.
n
show gatekeeper status Shows GK status (authorization and authentication) and zone status.
Problem Resolutions for Gatekeepers Even with the most detailed planning, step-by-step implementation, and quality checks, problems can arise at any time during the life of your network. I will assume that you are familiar with the Cisco ideology on troubleshooting. The same philosophy is at play here.
Is your GK working as expected? If your GK shows no endpoints registered or your endpoints cannot locate a GK, you need to check your GK configuration: 1. Make sure the GK service has been defined. 2. Check your GK and domain names for typos. 3. Issue no shutdown under the GK configuration.
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VOICE GATEWAYS
Voice Gateways The GW is probably one of the most important devices on a VoIP network. Although the use of a GW is an option on a network, it should be considered mandatory. If you do not use them, you really do not have a real network. You can implement GWs with just a few commands on the end devices; the majority of configurations are located on the GWs. Gateways act as the translator between different types of networks, dial planner, security enforcer, and connectivity director to the public switched telephone network (PSTN) and voice/data networks. Gateways come in analog or digital, depending on the requirements of your network. Cisco offers a few different platforms for GWs. Some support only a few connection lines; others support many connection lines. Gateways at the most basic level route calls to the PSTN or network depending on the destination of the call. The other functions GWs provide include the following: n
Call control and call routing
n
Call setup and teardown
Protocol selection is an important process. It must take into consideration the special needs of the network and how your GWs will function. The next major point to consider is the number of options that most of the commands offer to further fine-tune the network to the required specs. Imagine the protocol available as a pizzeria; there are many pizza places to order from. So, the choice depends on your needs. Does the place offer delivery or only take-out? Think of the options as if you are ordering a pizza. You can choose to add sardines, mushrooms, sausage, and anchovies (although you will probably have heartburn afterward). This is the same as with your voice network: adding or changing functions and options add to the heartburn. In the end, you have your pizza. The moral of this is that to ensure proper functionality, know exactly what you are adding to your network.
Protocols Used by Gateways Gateways can use a few of the main VoIP protocols on the scene today. Here are a few that we discuss: H.323, SIP, SCCP, and MGCP.
Complexity with H.323
The process of setting up a GW can be either extremely easy or very complex depending on what functionality the network needs. Your configuration can be as little as a few lines for an easy configuration, to a few hundred lines for the more complex configurations. All of this depends on the protocol your company decides to implement.
H.323 is a complex protocol from its inception. We need to face the facts; the protocol was built to accomplish a job. There are lesscomplex protocols (SIP, for instance). H.323 was built to use TCP/IP and take advantage of its robustness. The components in H.323 are terminals, GWs, multipoint control units (MCU), and GKs.
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VOICE GATEWAYS
IP
IP
Gateway Bragg
IP
IP
V
Chicago
NYC Gateway Bragg
IP
FIGURE 1.7
Gateway V Benning
V
Simple Gateway Life
In the preceding diagram, the GW is not connected to another GW or GK.
IP
FIGURE 1.8
GW-to-GW dating
This diagram shows a more complex GW network with a connection to another GW and a PBX.
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CCVP GWGK Quick Reference Sheets CCVP GWGK Quick Reference Sheets By Brion S. Washington ISBN: 1587053241 Prepared for Duc Le, Safari ID:
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VOICE GATEWAYS
As you can tell from these diagrams, you can use GWs in a variety of network layouts. The only difference is who the GW connects to, how many devices it is communicating with, and how many commands you need to configure to accomplish it. Keep in mind that these connections can be over a single interface, multiple interfaces, a single port, or multiple ports.
V V
V
Session Initiation Protocol IP V V IP IP
FIGURE 1.9
The Nightmare
Session Initiation Protocol (SIP) is a simple protocol that uses TCP or UDP. It is built on the client/server model. RFC 2543 defines SIP as a signaling protocol for creating, modifying, and terminating Internet multimedia or Internet telephony sessions. SIP uses multicast, unicast, or a combination of the two for communications. SIP can be used by other VoIP protocols to start, locate, and terminate calls. SIP can interoperate with other VoIP protocols on devices that allow this. All devices on this type of network are a client, server, or a proxy. The client (user agent client) sends the invites; the server (user agent server) provides the service and sends a response. Proxy servers (user agents) act like both a client and a server; it makes requests for others. These requests are handled by the proxy or sent to another server to fulfill the request.
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VOICE GATEWAYS
SIP URL addresses follow the user-at-host style (
[email protected] |
[email protected] |
[email protected]). The format is host @ network address or domain. SIP addresses must be unique. When a SIP client needs to locate a server, it does so by the server IP and port number in the Request URI or the locally configured proxy server. SIP can be used for connections to the PSTN, voicemail, and application servers. SIP offers a variety of services that many companies could find useful, including the following: n
Fax support
n
Call transfer
n
Endpoint locating
n
Conferencing
n
Media capability determinations
n
Dynamic endpoint registration
n
Toll bypass
SIP Call Flow
Setup INVITE Call Proceeding Setup 100 Trying Call Proceeding Alerting 180 Ringing Alerting
1-Way Voice
2-Way RTP Channel
1-Way Voice Connect
200 OK Connect Connect ACK ACK
2-Way Voice
2-Way RTP Channel
2-Way Voice Disconnect
BYE Disconnect Release 200 OK Release Complete Release Complete
FIGURE 1.10
SIP Call Flow
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VOICE GATEWAYS
Call steps using a single SIP redirect server:
SIP Server SIP
1. The calling party sends an invite to the redirect server.
IP UAC
2. The redirect server ask the location server to find the path for the
called party.
SIP
RTP
3. The redirect server sends information to the calling party. 4. The calling party sends an acknowledgment of receipt to the V
PSTN
redirect server.
PBX
5. The calling party sends a request to the called party.
FIGURE 1.11
Simple SIP Network
6. The called party responds. 7. An RTP session is finally opened.
Call steps on a single-proxy-server network: 1. The end user registers with a registrar server.
SIP Server A-1
SIP Server B-1
2. The call is initiated by caller. The request is sent via the SIP server
(proxy or redirect).
IP
IP
3. The server forwards an invite to the called party. 4. The called party responds. 5. The server forwards the called party’s response to the calling
party. V
V
User A
FIGURE 1.12
User B
Call Steps to a Remote Network
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VOICE GATEWAYS
SIP request types:
1. User A sends an invite to proxy server A-1.
The location server could also be used to locate User B.
n
ACK—Confirms the client has received a response to the INVITE request.
n
BYE—Terminates a call.
n
CANCEL—Cancels pending searches; call that is currently in progress continues.
4. User A and User B trade responses and acknowledgments.
n
INVITE—Invites to participate in a call.
5. An RTP session is established between User A and User B.
n
OPTIONS—Queries the capabilities of servers.
n
REGISTER—Registers the address with a server.
2. Proxy server A-1 contacts proxy server B-1. DNS is used by
proxy A-1. 3. Proxy server B-1 contacts user B.
The location server could also be used here.
Connection to a PSTN:
SIP works by using simple messages to communicate. Here are the messages:
1. All steps are repeated. 2. SIP GW and SIP endpoint create RTP session. 3. POTS signaling is translated into SIP messages.
The SIP message format:
n
1xx: Informational—Received request, processing the request.
n
2xx: Success—The action was successful.
n
3xx: Redirection—Further action needed.
n
4xx: Client Error—The request contains bad syntax or cannot be done on this server.
n
A start line
n
One or more header fields
n
An empty line
n
5xx: Server Error—The server failed to fulfill the request.
A message body (optional)
n
6xx: Global Failure—The request cannot be fulfilled at any server.
n
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OPEN SET TLEMENT PROTOCOL
Skinny Client Control Protocol Skinny Client Control Protocol (SCCP) is a Cisco proprietary VoIP protocol used for communications between Cisco CallManagers and IP phones. Skinny has a simple architecture; it uses TCP to communicate with the CallManager and UDP between the endpoints. Skinny has a reduced overhead because most of the processing happens on the CallManager. IP phones must be configured with the IP address of the default GW and the IP address of a TFTP server and must have an IP address for SCCP to work.
Media Gateway Control Protocol Media Gateway Control Protocol (MGCP) is a simple VoIP text protocol used to control GWs. RFC 2705 defines this protocol as a master/slave arch, where call control devices take a specific port on a GW under control. This protocol offers centralized GW administration. When centralized administration is used, you can realize large, scalable VoIP environments. If changes are made, they are made in one area rather than multiple areas. Therefore, most of your configuration is done on a few devices while others have bare configurations. Cisco CallManager uses MGCP on UDP port 2427 to communicate with GWs. MGCP, like other VoIP protocols, has components that make up the network. They are endpoints and call agents. Endpoints are voice ports on GWs controlled by MGCP. The next components are call agents. Call agents are the external control devices. The call agents contain all of your dial plans.
Open Settlement Protocol The European Telecommunications Standards Institute (ETSI) Project TIPHON (Telecommunications and Internet Protocol Harmonization Over Networks) Open Settlement Protocol (OSP) provides a way for VoIP service providers to use services provided by OSP clearinghouses. This client/server protocol is used to exchange accounting, authorization, and usage information. The purpose of OSP is to manage access control, accounting, and interdomain routing. You can use this protocol between H.323 GWs, H.323 GKs, softswitches, and SIP proxies. Public Key Infrastructure (PKI) is used to provide secure services. The server creates a secure link for the transfer of information. When the link is established, a secure token-based signature is used to certify that the call is authorized. Clearinghouses provide best route and least cost path selections based on subscriber requirements.
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GET TING A GATEWAY UP AND RUNNING WITH H.323
Getting a Gateway Up and Running with H.323 To set up a functional GW with GK support, follow these steps: 1. Set up GK and GW registration:
CCVP GWGK Quick Reference Sheets by Brion S. Washington
Example: Gateway Interface Configuration interface FastEthernet0/0 ip address 5.5.5.98 255.255.255.128 h323-gateway voip interface h323-gateway voip id Gk-Infantry ipaddr 75.75.50.1 1719 h323-gateway voip h323-id GW-1
GW1(config)# gateway
Configuring SIP User Agents
2. Create the H.323 interface: GW1(config-if)# h323-gateway voip configuration mode
interface **under interface
Setting up SIP is extremely easy: 1. Enable SIP.
3. Create the H.323 ID: GW1(config-if)#
2. Set up alternate SIP servers (optional).
h323-gateway voip h323-id
3. Set up dial peers.
This command assigns a unique H.323 ID for the device. This ID is used on the network for identification and connectivity.
sip-ua sip-server ipv4:75.1.1.69
4. Manually configure GK information (optional): GW1(config-if)# port
h323-gateway voip id (name) ipaddr
or IP address sip-server dns:my-sip.com
This command manually sets up connections to other control devices (GW or GK). You may leave out this command if you need only a basic GW.
Dial-peer voice 12 voip Session target sip-server
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ADVANCED GATEWAY OPTIONS MEDIA RESOURCES
Advanced Gateway Options Media Resources Digital signal processors (DSP) are the heart of any VoIP network. DSPs provide the most important job of anything on the network. If your network is set up 100 percent correctly, but you have a bad DSP located at the beginning or end of a voice path, all is for naught. DSPs are where the encoding and decoding of your voice between analog and digital devices takes place. All Cisco voice products have some sort of DSPs; without having this hardware, they would not be able to provide voice support. Cisco has an extensive list of products that can and do provide voice support. The main components are the voice cards. Voice cards are what hold the DSP and are made to fit any voice platform. There are quite a few types and configurations of cards. This guide does not cover any of the actual cards specifically; it uses general terms and hardware. The voice card naming convention will give you an idea of how many calls it can support, and which platform the card will work in. Let’s go over the resources that DSPs provide to Cisco voice networks.
CCVP GWGK Quick Reference Sheets by Brion S. Washington
DSP Codec Functions If the DSP is the heart of the network, it can be said that codecs are the blood. They encode and decode incoming analog signals into digital signals and vice versa. There are many codec flavors to choose from. With so many choices for codecs, there are bound to be tons of options for fine-tuning. Codec configuration can be divided into four groupings: global, port, voice card, and dial peer.
Voice Card Codec Options Because the voice card is where the DSPs are located, this seems like the logical place to start tuning resources to your requirements. Codec complexity is a compression technique that states the amount of processing required for voice compression. The higher the complexity, the fewer number of calls the DSP can process. When you are using high complexity, your voice channels available will decrease. Steps required: 1. Select the voice card. 2. Enter the codec complexity you want: GW-1(config) # voice-card 1 GW-1(config-voicecard) # codec complexity [high | medium | flex] ** Flex is available on C5510 DSP**
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ADVANCED GATEWAY OPTIONS MEDIA RESOURCES
CCVP GWGK Quick Reference Sheets by Brion S. Washington
The dial peer uses the codec command to specify which codec to use. The codec command is entered under dial-peer configuration. We go over dial peers in more detail later, so this is just to give you an idea of where the codec command fits:
Assign voice class to dial peer:
Dial-peer
GW-1# Show running-config **all output removed except voice class and dial peer
dial-peer voice 417001 voip
GW-1(config)# dial-peer voice 50 voip GW-1(config-dial-peer)# voice-class codec 1
destination-pattern 417[4,6-9].T voice class codec 1
codec g723r63
codec preference 1 g729r8
session target ipv4:18.82.16.12
codec preference 2 g729br8
You might also decide to just let the two parties involved negotiate the codec used during the H.245 negotiation process. If this is your preference, you may configure a global class to handle all the negotiations not specifically configured under individual dial peers.
codec preference 3 g723r63
The global command voice class codec enables you to configure the preference of codec once for any dial peers you want to associate:
codec preference 8 g711alaw
codec preference 4 g723r53 codec preference 5 g723ar63 codec preference 6 g723ar53 codec preference 7 g711ulaw ! !
GW-1(config)# voice class codec 1
dial-peer voice 50 voip
GW-1(config-class)# codec preference 1 g729r8
description My_house [402]
GW-1(config-class)# codec preference 2 g729br8
destination-pattern 402.T
GW-1(config-class)# codec preference 3 g723r63
voice-class codec 1
GW-1(config-class)# codec preference 4 g723r53
session target ras
GW-1(config-class)# codec preference 5 g723ar63
!
GW-1(config-class)# codec preference 6 g723ar53 GW-1(config-class)# codec preference 7 g711ulaw GW-1(config-class)# codec preference 8 g711alaw
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TRANSCODING AND CONFERENCING
Transcoding and Conferencing What would the world be like without the ability to talk to multiple people simultaneously? Would the United Nations General Assembly be able to resolve global problems if there were no translators on staff? Well, the same can be said about VoIP communications. Media resources is the term for DSPs responsible for providing the resources to handle conferencing and transcoding. The voice GW uses SCCP to communicate with the Cisco CallManager, which controls the DSP farm.
Configuring Conferencing and Transcoding This section covers how to configure the two likely DSP types you will find on the exam. The two types are C549, found on Network Modules (NM), or the C5510, found on the Packet Voice DSP Modules (PVDM). NM-HDV (C549) setup:
GW-1(config)# sccp local interface
2. Set up the CallManager IP:
5. Set the number of sessions: GW-1(config)# dspfarm confbridge maximum sessions number GW-1(config)# dspfarm transcoder maximum sessions number
6. Enable DSP farming: GW-1(config)# dspfarm
7. Set up the resources for conferencing and transcoding on
CallManager.
Setting Up Enhanced Conferencing and Transcoding (C5510)
1. Set up the SCCP interface:
GW-1(config)# sccp ccm ipaddr [priority priority-level]
3. Enable SCCP:
sessions: GW-1(config-voicecard)# dsp services dspfarm
Enhanced conferencing and transcoding enables you to create multiple profiles. These profiles allow control of resources and let the GW register with multiple CallManager groups.
1. Set up the SCCP interface:
GW-1(config)# sccp
4. Configure the voice card for conferencing and transcoding
GW-1(config)# sccp local interface
2. Set up the CallManager IP: GW-1(config)# sccp ccm ipaddr identifier identifier [version ccm version]
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TRANSCODING AND CONFERENCING
CCVP GWGK Quick Reference Sheets by Brion S. Washington
GW-1(config-dsp-profile)# associate application sccp
3. Initialize SCCP:
GW-1(config-dsp-profile)# maximum sessions {hardware | software} sessions
GW-1(config)# sccp
4. Configure the voice card for conferencing and transcoding
sessions:
GW-1(config-dsp-profile)# no shutdown
8. Associate profiles with CallManager groups:
GW-1(config)# voice-card 0
GW-1(config)# sccp ccm group number
GW-1(config-voicecard)# dsp services dspfarm
GW-1(config-sccp-ccm)# associate ccm identifier
priority priority
GW-1(config-sccp-ccm)# associate profile profile register devicename
5. Set up the DSP farm profile for transcoding: GW-1(config)# dspfarm profile profile transcode
**profile is a number range of 1 to 65535
**profile is a number range of 1 to 65535
GW-1(config-sccp-ccm)# bind interface interface
GW-1(config-dsp-profile)#associate application sccp
9. Set up resources in CallManager.
GW-1(config-dsp-profile)#maximum sessions number GW-1(config-dsp-profile)#no shutdown
6. Set up the DSP farm profile for conferencing: GW-1(config)# dspfarm profile profile conference **profile is a number range of 1 to 65535 GW-1(config-dsp-profile)#associate application sccp GW-1(config-dsp-profile)#maximum sessions number GW-1(config-dsp-profile)#no shutdown
7. Set up the DSP farm profile for MTP: GW-1(config)# dspfarm profile profile mtp **profile is a number range of 1 to 65535
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DROP AND INSERT MULTIPLEXING
Drop and Insert Multiplexing The needs of modern businesses have made advances in options that networks provide. Some come in the form of protocols, others in the form of hardware. One of these is drop and insert multiplexing. Drop and insert hardware allows companies to buy a single network module. These modules support data and voice. They include an integrated channel service unit / data service unit (CSU/DSU), and some newer models echo cancellation modules. Voice support can be used only for DS0 or PRI modes. Now with a Voice WAN Interface Card (VWIC), Cisco devices can offer a better value for companies by requiring only a single module to be purchased. When they install the VWICs, they can do both data and voice simultaneously. The VWIC can help prevent issues from arising. After the module is in, there is no need to pull the module out to adapt to network changes.
Example: Drop and Insert for T1 GW-3# show run Building configuration... card type t1 0 0
**configures card type
! no network-clock-participate wic 0
**VWIC won’t use network timing
! controller t1 0/0 framing esf linecode b8zs channel-group 1 timeslots 1-12 speed 64
**specify group slots
tdm-group 2 timeslots 13-24 ! controller t1 0/1 framing esf linecode b8zs channel-group 1 timeslots 1-12 speed 64 tdm-group 2 timeslots 13-24 ! ! interface Serial0/0:1 ip address 75.75.0.10 255.255.255.0 ! interface Serial0/1:1 ip address 75.75.0.11 255.255.255.0 ! connect t1-xconnect T1 0/0 2 T1 0/1 2 Insert between ports
**configures Drop and
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ANALOG CONNECTIVIT Y
Analog Connectivity
E&M
Since the day the telephone was invented, analog has been the method used by phones. Analog waves are converted by the phone into electrical signal to cross the network; it is converted back to analog at the destination. Analog has been around for a long time. There are a few technologies that have been born in the analog communication world, as discussed in the following subsections.
Most people in the telephony world have heard of Earth and Magnet, or Ear and Mouth (E&M). E&M is used in analog trunking. These are the connections between central office switches and private branch exchange (PBX), or for inter-PBX. E&M performs the on-hook and off-hook signaling to confirm when the handset is picked up or put down. Signaling of the handle’s on-hook and off-hook helps reduce the chance of two people picking up the phone at the same time. E&M comes in five variations, as described in the following table.
Analog Signaling There are a few choices for analog signaling, depending on what you require. Signaling is the call control messages sent between devices on the PSTN. Signals include information such as the phone is ringing, incoming call, line is busy, and so on. Some of the older signaling types have been mostly replaced be newer, more efficient signaling. An important part to remember is that analog uses CAS (Channel Associated Signaling). CAS sends the control signaling on the same channel as the data. We discuss the types of signaling you can use and how to get them working on Cisco GWs throughout the rest of this section.
Type
Information
On-Hook
Off-Hook
1
Used in the United States
M lead connected to ground E lead open
E lead to ground M lead to battery
2
Used in the United States
E and M lead open
M lead connected to CO E lead to PBX ground
3
Older signaling
M lead to CO ground E lead open
M lead to CO battery E lead to ground
4
Not supported by Cisco
N/A
N/A
5
Used in the United States and Europe
E and M open
M lead to PBX ground E lead to ground
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ANALOG CONNECTIVIT Y
E&M signaling connect PBXs using eight-wire circuits. The breakdown of the wire is four wires for signaling and two or four wires for the voice. Each wire has a different responsibility, as shown in the following table.
Setting Up E&M Trunking E&M ports connect to PBXs, so you must configure the port to match the PBX brand you are using:
Lead
Description
Pin Number
n
Call progress tone
SB
-48V signaling battery
1
n
Impedance
M
Signaling input
2
n
Operation
R
Ring, Audio Input
3
n
Signal type
R1
Ring, Audio Input / Output
4
T1
Tip, Audio Input / Output
5
n
Type
T
Tip, Audio Input
6
Select voice port:
E
Signaling Output
7
GW-2(config)# voice-port
SG
Signaling Ground
8
nm-module/vic-module/port-number
Voice-port 2/1/1
Configure signaling: GW-2(config-voiceport)# signaling [delay-dial | immediate | winkstart]
Configure interface type: GW-2(config-voiceport) # type
[1 | 2 | 3 | 5]
** Type 4 is not supported on Cisco devices
Configure country code: GW-2(config-voiceport) # country
country code
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ANALOG CONNECTIVIT Y
Configure cabling to use:
dial-type Type of out signaling to use for port.
GW-2(config-voiceport) # operation [ 2 wire | 4 wire]
dial-type [dtmf | pulse | mf]
ring number Number of rings.
Configure PBX termination impedance: GW-2(config-voiceport) # impedance [600c | 600r | 900c | complex1 | complex2]
ring number number
signal Type of signaling for port. FXS and FXO ports signal [loop-start | ground-start]
Voice Ports The most basic connection type is the port. Without these ports, no communications would take place. Ports provide connection to analog and digital devices. These devices can be a telephone, IP phone, fax, PBX, and so on. The ports you need to be familiar with are FXO, FXS, and DID. The FXO and FXS are the ports used in VoIP. The FXS (Foreign Exchange Station) is what your analog devices are connected to. FXS ports must supply voltage, ring tone, and power. The FXO (Foreign Exchange Office) is connected to the PSTN.
FXO ports connect GWs to the PBX. These connections to the port act like telephones connected to the PBX. The commands used are as follows: description [description]
FXS Ports FXS ports connect your telephones and fax machines to your GW. These ports must provide the same functionality as PBXs. You can configure a few options for FXS ports, as follows: busyout Allows busyout of port.
FXO Ports
description Sets description for the port.
supervisory disconnect Enables supervisory disconnect signal. supervisory disconnect
busyout
cptone Sets call-progress tone for country. cptone [locale]
*** country location
description Sets description for port. description [description]
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ANALOG CONNECTIVIT Y
disconnect-ack Removes line power, and returns acknowledgment of disconnect signal. disconnect-ack
callerid Enables caller ID. callerid
connection Specifies connection type. connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]}
DID It allows calls to bypass an operator. When you call an office, you can either ask the human operator to connect you to the extension, or the PBX can bypass the operator and perform the task itself. The key points to remember about DID is Cisco offers limited support for DID functionality; only few modules support it. The next point is DID works on incoming calls only. Configure POTS dial peer: dial-peer voice 34 pots
ring cadence Sets ring cadence (cptone overrides this command). ring cadence {pattern-number | define pulse interval}
Set up direct inward dial: Direct-inward-dial
ring frequency Sets ring frequency. ring frequency [frequency]
Digital Connections to the PSTN
signal Type of signaling for port. FXS and FXO ports signal [loop-start | ground-start]
E&M ports signal {wink-start | immediate | delay-dial | lmr}
station id name Caller ID information sent to caller. station id name[name]
station id number Caller ID information sent to caller. station id number [number]
The voice gateway needs to interface with an assortment of devices over various connections. Gateways without connections are not of much use to a network. The connections can range from small trunks’ DS0 to DS-3. All the connections have the same purpose: to provide a travel medium and connectivity to other devices. Cisco GWs can be outfitted with a variety of modules and cards for connection to almost any type of network connection. The main connections you must have knowledge of and be able to configure are as follows: n
ISDN PRI
n
ISDN BRI
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ANALOG CONNECTIVIT Y
n
Analog
4. Configure the IP address.
n
CAS
5. Configure SPIDs (optional).
n
Drop and Insert
6. Associate the dial list. GW-1(config)# isdn switch-type basic-5ess GW-1(config)# dialer-list 2 protocol ip permit
ISDN
GW-1(config)# int bri 0
Integrated Services Digital Network (ISDN) is a digital transport technology used to connect small office, home office (SOHO), or companies to the PSTN. ISDN is a circuit switching technology composed of two types of channels. The Bearer channel (B channel) is the channel that data uses to move across the network. The Data channel (D channel) is responsible for transporting control protocols. The method of sending call control messages over a different channel than the D channel is known as Common Channel Signaling (CCS). ISDN comes in two configurations: Primary Rate Interface (PRI) and Basic Rate Interface (BRI). PRI is a DS1 that has a capacity of 1.544 Mbps for T1 and 2.048 Mbps for E1; BRI has a capacity of between 54 kbps and 128 kbps.
GW-1(config-if)# ip address 5.5.6.12 255.255.255.128 GW-1(config-if)# isdn spid1 74798964631312 8884710 GW-1(config-if)# isdn spid1 74798964631315 8884714 GW-1(config-if)# dialer-group 2
PRI Configuration 1. Configure the ISDN switch type. 2. Configure the controller (T1-E1). 3. Configure the D channel. GW-1(config)# isdn switch-type primary-5ess GW-1(config)# controller T1 1
BRI Configuration 1. Configure the ISDN switch type. 2. Create a dial map.
GW-1(config-controller)#
framing esf
GW-1(config-controller)# clock source line secondary 1 GW-1(config-controller)#
linecode b8zs
GW-1(config-controller)# pri-group timeslots 1-24
3. Configure the BRI.
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ANALOG CONNECTIVIT Y
D-Channel Configuration
CAS Configuration
GW-1(config)# interface Serial1:23
Channel associated signaling is sent over the same channels as the data/voice. It is used primarily with PBXs over E1 or T1 lines. CAS is also known as robbed-bit signaling, because some of the bits used to carry data information are robbed for signaling purposes. Although some bits are stolen for signaling, all channels are still available for carrying the data.
GW-1(config-if)# no ip address GW-1(config-if)# isdn switch-type primary-5ess GW-1(config-if)# isdn incoming-voice modem GW-1(config-if)# isdn send-alerting
Now that we have ISDN BRI and PRI set up, we turn our attention to verification and troubleshooting. You can use some good commands to help verify your configuration, or to resolve issues that pop up. One extremely vital area of ISDN is the cause codes. ISDN cause codes are present in numerous voice and ISDN command outputs. You can find an extensive list of ISDN cause codes at www.cisco.com. The show isdn status command displays Layer 1 through Layer 3 information. If status is not active, you need to check your connection. Also, check to make sure MULTIPLE_FRAME_ESTABLISHED is present.
GW-5(config)# controller T1 0 GW-5(config-controller)# framing esf GW-5(config-controller)# clock source line primary GW-5(config-controller)# linecode b8zs GW-5(config-controller)#ds0-group 0 timeslots 1-24 type fxo-groundstart
Syntax: ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-winkstart | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loopstart}
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CIRCUIT EMUL ATION
Circuit Emulation
or
Circuit Emulation over IP (CEoIP) is an end-to-end circuit emulation service for time-sensitive data over an IP network. This transparent transport is used by applications that need a constant bit rate. The endpoints can use adaptive clocking to sync the clocks. CEoIP has a few optional features, such as CAS, data protection, and payload compression.
GW-5 (config-controller)# framing {crc4 | no-crc4 | unframed} framing
**E1
GW-5 (config-controller)# clock source {internal | line | adaptive channel-number} GW-5 (config-controller)# cablelength {long | short} {attenuation | length} GW-5 (config-controller)# crc-threshold value GW-5 (config-controller)# description text
When and why you should use CEoIP:
GW-5 (config-controller)# loopback{local {line | payload} | network}
n
Automated industrial environments
n
Crypto tunneling
n
Legacy services and applications
GW-5 (config-controller)# timeslots timeslot
n
Radar data
[speed {56 | 64}]}
n
Satellite data
Configuring the connection using xconnect:
Creating CEM channels on the T1/E1 line: cem-group group-number {unframed |
GW-5(config)# cem slot/port/channel xconnect remote-ip-address virtual-connect-ID
Configuring CEoIP
GW-5(config-cem)# encapsulation
Configuring the NM-CEM-4TE1 card:
encapsulation-type
GW-5# configure terminal
GW-5(config-cem-xconnect)# local udp port port-number
GW-5(config-cem-xconnect)# local ip address ip-address
GW-5(config) # card type {t1 | e1} slot
GW-5(config-cem-xconnect)# remote udp port port-number
Configuring the T1 /E1: GW-5 (config) # controller {t1 | e1} slot/port GW-5 (config-controller)# framing {esf | sf | unframed} framing
**T1
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FA XING
Faxing
Modem Pass-Through
The same reasons for running voice over data are the same reasons for faxing over data. Although faxing probably does not take as long to complete as a phone conversation, expenses can be high for longdistance faxing. There are a few ways to get that important fax across a voice network. Each of them was developed for specific situations.
This method is like fax pass-through except there are modems at both ends of the connection. Modem pass-through uses the same transparent connection, and G.711, with no VAD. The supported protocols are H.323, MGCP, and SIP.
Modem Relay Cisco Fax Relay This Cisco proprietary protocol was created to send faxes over a VoXX network. The DSP acts like a virtual fax machine, converts the signal back to its original form, and then sends it to the remote fax machine. To save bandwidth, it uses 9.6 kbps. A key point to know is why a fax cannot be treated like a normal call. When a fax comes into the GW, it cannot be treated like a normal voice call, because otherwise the GW will miss the fax signaling and continue to treat the call as a normal voice call.
Fax Pass-Through This method uses a transparent IP connection to transport the fax. In this mode, no conversions take place. It treats the fax like a voice call with no special fax capabilities. The codec used is G.711, with no Voice Activity Detection (VAD), and supports H.323, MGCP, and SIP.
Modem relay is a method built to deal with overhead. It uses Simple Packet Relay Transport (SPRT) to transport modem signals over IP. The SPRT GW converts analog to digital. SPRT uses the UDP protocol, and is less sensitive to jitter and clocking issues than modem pass-through. The supported protocols are H.323, MGCP, and SIP.
T.37 Fax Store and Forward T.37 is the way to send faxes across your network like email attachments. The document is scanned, converted to TIFF, and then sent to a receiver’s email address using Simple Mail Transport Protocol (SMTP). The Cisco way of implementing T.37 is by way of GWs. On-ramp GWs do the conversions and sending of faxes; the off-ramp GWs are where the messages are received or held until forwarded. T.37 uses Multi-Media over IP (MMoIP) dial peers. Interactive Voice Response (IVR) applications are used for on-ramp and off-ramp functions.
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TCL SCRIPTS
T.38 Fax Relay Like all things in life, there is an industry standard, and then there is the Cisco way. Well, T.38 is the industry standard for fax relay. The two ways are similar, but T.38 needs to have a T.38 GW on both ends of the connection. The main difference is that T.38 allows faxing to virtual fax machines compatible with T.38. This protocol uses VoIP dial peers. The supported protocols are H.323, MGCP, and SIP.
Tcl Scripts Tool Command Language (Tcl) is an interpreted scripting language. This means that you do not need to compile anything to use it. The language provides a set of commands that give the basic functions. If more advanced or specialized functions are required, you can use extension sets to augment commands. Cisco has created an extension set for IVR functionality. A Tcl IVR script is activated when a call comes into the GW.
You can accomplish a lot of things when you have access to a scripting language. On the GWGK exam, you’ll be able to narrow down the extensive list of things you’re able to do and concentrate on only a few things. What you need to do for your exam is as follows: n
Download Tcl scripts.
n
Configure Auto Attendant.
n
Describe usage (billing, radius, AAA).
Download Tcl Downloading the Tcl software is fairly simple. You need to make sure your Cisco IOS release supports this functionality. Next, you need a CCO login to download Tcl from http://www.cisco.com: http://www.cisco.com/cgi-bin/tablebuild.pl/tclware http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp
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TCL SCRIPTS
Get scripts from a TFTP server:
3. Choose a name and pilot number to use by every AA application.
archive tar /xtract tftp://75.75.10.240/cme-mytcl.tar flash:
4. Choose the number and type of call routing (menu) options. 5. Customize your prompts.
Auto Attendant
6. Set up call-retry parameters.
Auto Attendant is configuring your CallManager or CallManager Express to auto-answer incoming calls. When a call comes in, the Auto Attendant can provide a greeting, followed by prompts to help direct the call to the right group. You can also use Auto Attendant to provide information and help to begin problem resolution. You configure Auto Attendant in a three-step process:
7. Choose where to send unanswered calls or full-queue calls.
1. Plan the call flow.
Download the Audio Files and Tcl Scripts 1. Download files to a TFTP server accessible by CallManager. 2. Uncompress files in privilege mode: GW-4# archive tar /xtract tftp://75.75.100.240/cmeyourscript.tar flash:
2. Download the audio files and Tcl scripts. 3. Configure incoming dial peers.
Configure Dial Peers Call Flow Design
dial-peer voice 44 pots
You must determine how to handle incoming calls. When calls come in, how many queues do you have? You must make routing decisions. You must consider options to offer callers.
service my_cool_script incoming called-number 40255511 port 1/1/0
1. Choose a name for the call-queue service. 2. Decide how many authentication, authorization (AA) applications
are needed.
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IP-TO-IP GATEWAY (IPIPGW )
IP-to-IP Gateway (IPIPGW) The main points you need to understand about IPIPGWs are the services they provide for your network and the key reasons they are not considered normal voice GWs. IPIPGWs offer services such as video support, interprotocol networking, address hiding, and better billing capabilities. Billing can be simplified with the addition of IPIPGWs. Sometimes you need to bypass the PSTN and send VoIP between two VoIP networks. There are many names used to refer to this type of device: IP-to-IP gateway, border element, and session border controller. Wholesale VoIP resellers (WVR) or Internet telephony service providers (ITSP) use this method to connect to other VoIP resellers. One of the main reasons for implementing this kind of connection is for billing. When a WVR terminates calls between two other WVRs, the middleman misses out on billing opportunities because it does not control the terminating PSTN GW. The originating and terminating GWs see the call setup and release messages; the GWs in the middle do not. Both WVRs and ITSPs basically do the same thing, kind of like “tomayto or tomahto.” It’s all in how you perceive them. IPIPGW can hide IP addresses of your service providers from customers, making your network look larger than it actually is. Address hiding can be a useful tool to expand your company’s size without actually buying more equipment. The reason I say this is your company can now offer more destinations to clients without the fear of clients acquiring the destinations from your service provider. This type of address hiding is widely used by wholesale VoIP companies.
IPIPGWs are used like other voice GWs in providing interprotocol networking. With interprotocol networking, you translate one type of VoIP protocol used by the calling party into a different protocol used by the called party. IPIPGWs work with H.323 or SIP and require special Cisco IOS (IPIPGW Image) software to perform these functions. Another functionality of these types of GWs is the capability to support video calls. Video codecs supported are Far-End Camera Control (FECC), H.261, H.263, and H.264. The following figure shows an example and the likely steps used to provide IPIPGW in a real-world VoIP network. IPIPGW Company A
V Chicago
V Company Z
FIGURE 1.13
V Company D
WVR Example
1. Company A is a small company of five employees. They would
like to enter the wholesale VoIP / ITSP market. Company A buys a Cisco MultiService Gateway. 2. Company A buys a route from Company D to Chicago. 3. Company A is assigned a tech prefix of 333313# by Company D.
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IP-TO-IP GATEWAY (IPIPGW )
4. Company A sells VoIP minutes to Company Z for calls to Chicago. 5. Company Z is assigned a tech prefix of 55# by Company A. 6. Company Z has end users (callers) with family calling the Windy
City. Calls are sent to Company A’s IPIPGW IP address with a tech prefix of 55#. 7. Company A’s IPIPGW strips off 55# and forwards digits with
IPIPGW H.323 or SIP Setup 1. Enter voice services configuration: GW-1(config)# voice services voip
2. Specify allowed connections: GW-1(config-voi-serv)# allow connections [h323 | sip] to [h323 | sip]
333313# appended to Company D’s IPIPGW. 8. Call detail records (CDR) are checked on all minutes from
Company A’s IPIPGW for billing.
By default, H.323-to-H.323 connections are disabled. Any-to-POTS and POTS-to-any are enabled by default.
You can repeat Steps 2 through 8 as many times as necessary for each new route or company you add. The only thing that will stay constant is the tech prefixes assigned. The tech prefixes are assigned only once, when the first route is added, and remain the same throughout the business relationship. So, Company A now looks like a big WVR or ITSP, without the need to have equipment located in every destination they want to offer. There are advantages and disadvantages to this style of business. Advantages
Disadvantages
Centralized billing
No control of remote networks
Centralized security
Limited quality of service support
Low investment to start up
Test every new route
Unlimited growth potential
Constantly changing providers
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ADVANCED IPIPGW COMMANDS
Advanced IPIPGW Commands
voice class codec 1
Like any other technology, you can get up and running with some basic configurations, but you always have other options to help tweak your network. Following here are some of the options available for IPIPGW.
codec preference 2 g729br8
codec preference 1 g729r8 codec preference 3 g723r63 codec preference 4 g723r53 codec preference 5 g723ar63
Codec transparency is used to pass codec capabilities between endpoints. This command is the same as for normal GWs.
!
2. Set the preferences of codecs.
! dial-peer voice 50 voip
3. Assign the voice class to dial peers.
description My_house [402] destination-pattern 402.T
GW-1(config)# voice class codec 1
voice-class codec 1
GW-1(config-class)# codec preference 1 g729r8
session target ras
GW-1(config-class)# codec preference 2 g729br8
!
GW-1(config-class)# codec preference 3 g723r63 GW-1(config-class)# codec preference 4 g723r53 GW-1(config-class)# codec preference 5 g723ar63 GW-1(config-class)# codec preference 6 g723ar53 GW-1(config-class)# codec preference 7 g711ulaw GW-1(config-class)# codec preference 8 g711alaw
GW-1(config)# dial-peer voice 50 voip
codec preference 7 g711ulaw codec preference 8 g711alaw
1. Configure the voice class.
Assign the voice class to dial peers:
codec preference 6 g723ar53
You can use the same scenario with a GK. When a GK is used, the only differences are new commands and terminology. Via zones are zones that have IPIPGW and via zone-enabled GKs. Via zone-enabled GKs send transient traffic to the IPIPGW. The IPIPGW terminates the call and then re-originates the call to the destination. Because the middleman can now see the call setup and call release, calls that normally would have been unbilled or not allowed are completed and billed.
GW-1(config-dial-peer)# voice-class codec 1
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ADVANCED IPIPGW COMMANDS
Company A
Via Zone Gatekeeper Setup
Gatekeeper
V
Local zone:
V Chicago
1. Enter the GK configuration: Gk-Calvary (config) # gatekeeper
V Company Z
FIGURE 1.14
V Company D
2. Configure the local zone: Gk-Calvary(config-gk)# zone local cope mydomain.com 75.75.86.97 11719 invia Gk-Ranger outvia Gk-Ranger
Via Zone Gatekeeper Example
zone remote zone-name domain-name ip-address gatekeeper name | outvia gatekeeper name
Note This example is a basic via zone topology. I did not place the remote GKs in here to help simplify the theory.
[port number][invia
Remote zone: 1. Enter the GK configuration: Gk-Calvary (config) # gatekeeper
2. Configure the remote zone: Gk-Calvary (config-gk)# zone remote pepsi 11719 invia Gk-Rebel outvia Gk-Rebel
mydomain.com 75.75.86.91
zone remote zone-name domain-name ip-address gatekeeper name | outvia gatekeeper name
[port number][invia
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GATEWAY REDUNDANCY
Advanced Gateways Options
Gateway Redundancy
The ability to fine-tune the network can lead to better resource usage, help reduce chances of errors, and aid in problem solving. You can configure literally hundreds of commands for a GW. You do not need to know all of them for the test, but you do need to know them for realworld applications. I will go over a few of the commands I believe are important for the GWGK exam:
Gateways also need to be configured with redundancy in case the primary GK or CallManager fails to diminish issues from arising. We cover three ways to accomplish this: Alternate GK, HSRP, and Redundant CallManager.
GW1(config)# Voice service [voip | voatm | pots | vofr] **used to apply voice services globally*** GW1(config-voi-serv)# Allow connections from-type [any | h323 | sip | pots] to to-type [any | h323 | sip | pots] ** this option under voice service command decides which VoIP connections the GW will allow. Use the No form to disallow.
Alternate Gatekeepers A GW can be configured with up to two alternate GKs: Interface FastEthernet0/0 Ip address 5.5.5.98 255.255.255.128 h323-gateway voip interface h323-gateway voip id Gk-Infantry ipaddr 75.75.50.1 1719 h323-gateway voip id Gk-Calvary ipaddr 75.75.66.1 1719 h323-gateway voip h323-id GW-1
Note Alternate GK is here because the configuration is done on the GW and not the GK. It can be argued that it can also go in the GK section.
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SUR VIVABLE REMOTE SITE TELEPHONY
Redundancy to CallManager Clusters You must configure a dial peer for each CallManager in the cluster you want to use. Remember that the preference command is important to the order in which they are selected. Dial-peer voice 199 voip Session target ipv4:192.168.1.11 Preference 1 Dial-peer voice 200 voip Session target ipv4:192.168.1.12 Preference 2 Dial-peer voice 201 voip Session target ipv4:192.168.1.3 Preference 3
You need to repeat this for all CallManagers. The preference range is 0 to 10; 0 is the default.
CCVP GWGK Quick Reference Sheets by Brion S. Washington
Survivable Remote Site Telephony If you were to design a modern VoIP network, your client would probably ask for a voicemail system, connections to the PSTN, remote sites, and so on. The first question you would be asked as the designer is what happens if we lose our network? How will we make and receive calls? Your response can be this: Because a Cisco solution was installed, there is no problem in the event of network failure. Survivable Remote Site Telephony (SRST) provides Cisco IP phones a backup path in the event of network failure, but only if they are connected to a local Cisco device. SRST makes sure your employees have at least basic phone service. When IP phones lose connectivity to all CallManagers, SRST uses the GW until connectivity to a CM can be reestablished. Your phones will kick into SRST in about two to three minutes. The phones must wait three times the keepalive period and try all your other CallManagers. In the event SRST failover occurs, your IP phones will periodically check whether a Cisco CallManager is available starting after two minutes of being in SRST mode. When communications have been reestablished with a Cisco CallManager, all phone service returns to normal. Here is what it looks like.
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SUR VIVABLE REMOTE SITE TELEPHONY
PSTN CM 1 IP
IP
to make sure that you’re running DHCP off your GW and not the CallManager. If your addresses are assigned by the CallManager, you will lose your lease. SIP requires that the GW act as a registrar server during SRST. It is also recommended that for faster failover, SIP should register with both the proxy and registrar server. SRST can be set up and will work if your network meets the following criteria:
V
V
SRST Router
CM 2
n
Cisco IP phones are connected to a Cisco device capable of SRST.
n
MGCP Fallback and SRST are both configured (MGCP only).
n
H.323 and SIP are getting call control and dial plans from the GW.
DHCP commands for the SRST router: IP
Central Office
Remote Office
ip dhcp pool pool-name
**Creates DHCP pool to assign IPs from**
network ip-address mask
prefix-length
option 150 ip ip-address
FIGURE 1.15
default-router ip-address
SRST
When SRST is configured, different protocols respond differently to the loss of Cisco CallManager connectivity. You should pick the protocol that best fits your survivability needs. Each of the protocols has special requirements that you must address. An MGCP has its own form of SRST called MGCP Fallback. You can configure both of these features on the same device in later Cisco IOS releases. H.323 will not keep active calls up if there is no signal from a Cisco CallManager. The unofficial workaround to this is no h225 timeout keepalive. This might help to keep calls active in the event of a link failure. You need
**IP of DHCP pool**
**IP of TFTP server to download image** **IP of router phones are connected to**
SRST for H.323: GW1(config)#
call-manager-fallback
GW1(config-cm-fallback)# ip source address 192.168.1.90 GW1(config-cm-fallback)# max-ephones 10 ** states how many calls will be allowed. This value depends on the Cisco IOS version and platform you are running. GW1(config-cm-fallback)#max-dn 8 **states the number of directory numbers supported. This is also platform dependent. Check command reference guide for some max-dn.
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GATEWAY PROBLEM SOLVING
SIP SRST
SRST on MGCP Gateways
When SIP goes into SRST mode, phones registered with a proxy server can use the voice GW acting like a SIP registrar.
MGCP requires that both SRST and MGCP Fallback be configured. In addition, you must also create a dial peer with basic PSTN connectivity. GW1(config-)# ccm-manager fallback-mgcp
1. Specify allowed connections.
GW1(config)#
2. Set up the GW as registrar.
service [alternate | default] service-name location
GW1(config)# exit
GW1(config)# voice service voip GW1(config-voi-serv)# allow-connections h323 to sip GW1(config-voi-serv)# allow-connections sip to h323 GW1(config-voi-serv)# allow-connections sip to sip GW1(config-voi-serv)# sip
Gateway Problem Solving Verifying and resolving issues with GWs can be accomplished with some of the commands you are already familiar with. You can verify which GKs are configured by one of two commands.
GW1(config-serv-sip)# registrar server
H.323 SRST Reminders 1. You need to configure POTS dial peers to allow direct extension. 2. Auto Attendant is provided by Tcl scripts. 3. Voicemail during SRST must be planned. There are a few ways of
doing this. There are more things that must be considered for phone functionality during SRST. I picked these because of their high daily usage.
show interface Show the VoIP interface and its configuration. show gateway Shows the list of GKs configured, along with their priority values. The last few places to check are H.323 configurations on the GW. You will need to check the interface and that your GK information was entered correctly. Next, you can make sure that you have general network connectivity.
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DIAL PEERS
Dial Plans
dial-peer voice (#) MMoIP
Dial plans are an essential part of completing a call. They tell the GWs where to send the call and how to treat the digits dialed. Dial plans are created using a combination of commands to perform call routing. The main components of dial plans are dial peers and their subcommands. Dial peers are probably the most important aspect of your network configuration. They enable you to do almost anything you could think of to do with incoming or outgoing calls. The responsibility the dial plan carries means your dial plan should undergo an extensive design and planning process to fulfill all requirements and functions. Dial plans can be located on GWs, CallManagers, PBXs, and softswitches.
Dial peers come in a few types: MMoIP, POTS, Voice over ATM (VoATM), Voice over Frame Relay (VoFR), and VoIP. They allow the GW to provide connectivity to the different network topologies, the PSTN, or mail servers. Most VoIP protocols can use dial peers. They are H.323, SCCP, and SIP. The basic configuration on these dial peers are similar; they vary in only a few commands and wording of the commands. Dial peers are picked by the longest matching number dialed. All dial peers must be in the UP operational state to be considered for matching.
dial-peer voice (#) VoIP
dial-peer voice (#) VoATM no shutdown The number range (#) is from 1 to 2147483647. Each dial peer must have a unique number assigned to it. If you assign the same number to dial peers, the last configured dial peer overwrites/changes dial-peer parameters. MMoIP dial peers are used mainly for fax capabilities. SMTP messages are forwarded to a server. VoIP dial peers are used for connections on packet voice networks. Dial strings are mapped to an IP address or remote IP.
Dial Peers
Dial-peer syntax and types:
dial-peer voice (#) VoFR
POTS dial peers are used to send calls to telephony devices or to the PSTN. Dial strings are mapped to voice ports. VoFR dial peers define connections that will use FRF.11 on a Frame Relay network. Dial strings are mapped to a permanent virtual circuit (PVC) or a switched virtual circuit (SVC). VoATM dial peers are used with AAL5 over ATM networks. Dial strings are mapped to a virtual circuit (VC). Note Not all Cisco platforms support all five types of dial peers.
dial-peer voice (#) POTS © 2007 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 63 for more details.
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CALL LEGS
Valid and Invalid Dial Peers
Dial Peer Matching
All dial peers must be in operational state to be considered for matching. When designing your dial peers, you must know which subcommands can be used together without adversely affecting the dial peer. There are three combinations that can be used to place peers in operational state:
Understanding how dial peers are matched to call legs is essential to a properly functioning voice network. Let’s go over how a GW matches inbound and outbound dial peers. It is assumed that you are familiar with how telephony works, so no in-depth explanation is given.
n
An answer-address is configured.
n
Destination-pattern with session-target or a voice-port.
n
Incoming-called-number configured.
Inbound calls use the entire string and are checked against configured dial peers in the following order: 1. incoming-called-number (called number “DNIS” information
used). 2. answer-address (calling number “ANI” information used).
To make sure all your dial peers are in the operational state you want them in, enter show dial-peer voice summary command.
3. destination-pattern (calling number “ANI” information used). 4. voice-port (that has dial peer port configured).
Call Legs
Note
Calls processed by the GW are categorized in one of two groups: inbound call leg or outbound call leg. By knowing how the GW sees the call, you will be able to set up your dial peers according to the network requirements. All calls have two call legs, the inbound and outbound legs. Inbound legs are the calls coming into the device; outbound legs are calls sent from the GW. These two legs are used to match which dial peer to use for completion of the call. With the flexibility to match either call leg, the dial plan should be able to handle most dial scenarios with ease.
AS5xxx series GWs stop after Step 3. Dial peer 0 will be used, and the call will be treated like a modem call. You might want to store this tidbit in your memory for possible future problem solving.
Outbound dial peer matching is much easier to remember. The choices are limited to only two options. All outbound dial peers are matched on the destination-pattern. The POTS peer then sends the call out the configured port; the VoIP peer sends the call to the value specified in the session-target.
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CALL LEGS
Default Dial Peers
destination-pattern command
While we are on the subject of dial peer selection, let’s look at default dial peers. When no matches can be found, the GW has two choices: drop the call or try to complete the call any way that it can. That is where the default peers come into practice.
This command can decide which dial peer to use and the destination. The variable can be an E.164 or user-defined number. The string can consist of 0 through 9, A through D, and a few special characters.
Default POTS dial peer configuration: n
No IVR applications
Default VoIP dial peer configuration:
Special characters: n
The asterisk (*) and pound sign (#) are on touchtone dial pads.
n
Backslash symbol (\), followed by a single character, matches the character.
n
Brackets ([ ]). A range is a sequence of characters in the brackets; only 0 to 9 are allowed.
n
Any codec
n
Fax-rate voice
n
Circumflex (ˆ), a match to the beginning of the string.
n
IP precedence 0
n
Comma (,), places a pause between digits.
n
No Resource Reservation Protocol (RSVP)
n
Dollar sign ($). Null string at the end of the input string.
n
VAD enabled
n
Parentheses (( )). A pattern that is the same as the regular expression rule.
n
Percent sign (%). Preceding digit occurred zero or more times; similar to wildcard usage.
n
Period (.). Wildcard matches any digits entered.
n
Plus sign (+). Preceding digit occurred one or more times.
n
Question mark (?). Preceding digit occurred zero or one time.
n
(T), variable digit string. Router will wait until all digits are entered.
An easy way to remember this is that the default values can be used on default dial peers, and you must sacrifice all the bells and whistles to complete a generic call.
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DIGIT MANIPUL ATION
answer-address command Identifies the incoming dial peer associated with the VoIP call. POTS calls are associated with the port the call came in on: n
The asterisk (*) and pound sign (#) are on touchtone dial pads.
n
Comma (,) places a pause between digits.
n
Period (.). Wildcard matches any digits entered
n
(T), variable digit string. Router will wait until all digits are entered.
Hunt Groups Hunt groups are used to add redundancy or routing choices to dial plans. They work by grouping dial peers that can complete specific calls into a hunt group. The commands to configure hunt groups and assign dial peers are as follows: hunt group (configured in global configuration mode). hunt group (configured under each dial peer you want to assign to a hunt group). huntstop (stops dial peer hunting if call fails).
dtmf-relay command How H.323 and SIP GWs relay dual-tone multifrequency (DTMF) tones between the telephony interface and the IP network.
Digit Manipulation
Syntax: dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal] [rtp-nte] [sip-notify]
Digit manipulation is something that you will definitely use at some point in your network. The idea behind this is pretty straightforward. Your GW receives digits from an outside source, and you want to do one of the following to the received digits:
progress-ind command This command allows progress indicators to be sent during a call to help with signaling. Syntax: progress_ind {setup | connect | progress | alert} {enable pinumber | disable}
n
Change digits
n
Add digits
n
Subtract digits
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VOICE TRANSL ATION RULES AND PROFILES
These are the main ways to manipulate digits. We manipulate digits for various reasons (for example, virtual private networks [VPNs], least cost routing, path selection, billing, and accounting). Manipulation of digits can make calling easier for the end users. Users can enter a few digits, and the GW can change the digits according to administrator wishes, but will add to the complexity of the configurations. That is the trade-off you must make.
Commands for Digits Manipulation n
Voice Translation Rules and Profiles Voice translation rules and profiles provide a powerful way to translate digits. The translation rules allows the called number and dialed number identification services (DNIS) digits to be manipulated. Voice translation profiles use voice translation rules and can change everything the rules can, plus they allow the redirecting number to be changed.
digit-strip
Used only on POTS dial peers only. When a call comes in on a POTS dial peer, the matched numbers are stripped, and the remaining numbers are forwarded on. Configured under POTS dial peer, this command is enabled by default. Usage: dial-peer voice 300 pots
Voice Translation Rule Wildcard patterns are used to help augment the pattern matching. .
Any single digit.
0 to 9, *, # Any specific character. [0-9]
Any range or sequence of characters.
*
Match none or more occurrences.
+
Match one or more occurrences.
?
Match none or one occurrence.
digit-strip
n
num-exp
Sometimes companies want to make dialing numbers easier for customers or employees. The num-exp command can take dialed digits and translate them into a different length of digits.
Syntax: rule precedence /match pattern/ /replacement pattern/
Examples:
voice translation-rule 1
num-exp
402……
1402……
num-exp 17142638080 8080
**1 is added to beginning
Example: rule 1 /^9...#809/ /809/
**incoming number changed to 8080
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VOICE TRANSL ATION RULES AND PROFILES
3. Apply profile:
rule 2 /^9...#1809/ /809/ rule 3 /^35681809/ /809/ rule 4 /^9...#1/ /1/
n
Globally
rule 5 /^35681876/ /876/
n
Dial peer
n
Interface
Cisco offers a test command to see whether your rule is working correctly:
n
Source IP group
n
Trunk group
test voice translation-rule name-tag input-number
n
Voice port
rule 6 /^3568011/ // rule 7 /^9...#/ //
Global example:
Voice Translation Profile
voice translation-profile VOIP-out
When you use this, you need to follow three steps:
translate called 1
1. Create at least one voice translation rule. n
Dial peer example:
128 rules max, 15 translations per rule
dial-peer voice 9003 voip
2. Create at least one voice translation profile and associated rules to
incoming called-number 9003#.T translation-profile incoming VOIP-out
the profile:
voice-class codec 1
n
1000 profiles supported
n
Unique name for each profile
n
You can apply one rule for each (calling numbers, called numbers, and redirected called numbers).
dtmf-relay h245-signal h245-alphanumeric
© 2007 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 63 for more details.
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COR
COR
Dial-peer cor list InternationalAccess Member International
Class Of Restriction (COR) enables you to restrict incoming and outgoing calls. 1. Define the COR label.
dial-peer voice 1 pots corlist outgoing LocalAccess
2. Build the permission groups.
destination-pattern [2-9]..........
3. Apply the COR list to dial peers:
port 1/0
a. Outbound dial peers, use corlist outgoing. b. Inbound dial peers, use corlist incoming.
! dial-peer voice 20 pots corlist outgoing LongAccess destination-pattern 1........... port 1/0
Define the COR Label Dial-peer cor custom Name local Name LongDistance Name international
! dial-peer voice 30 pots corlist outgoing InternationalAccess destination-pattern 0......... port 1/0 !
Build Permission Groups Dial-peer cor list LocalAccess Member Local Dial-peer cor list LongAcess Member LongDistance
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ENTERPRISE VERSUS SER VICE PROVIDER
CCVP GWGK Quick Reference Sheets by Brion S. Washington
Enterprise Versus Service Provider Two main types of VoIP networks are in use today: enterprise and service provider. Each has similarities with the other, and each has differences. Enterprise networks are built by companies trying to cut costs and leverage VoIP to help accomplish their business goals. Toll bypass is one of the leading factors for these kinds of networks. The layout of the networks can range from a single site to multiple sites, from centralized control to distributed control. Enterprise networks usually use Internet service provider circuits to complete their calls. They only go out on the PSTN when in failover mode or after the call has reached a switch local to the destination. Most of the equipment is owned by the company, giving the company administrative authority over the devices used. Service provider networks are networks built to provide a service to others for profit. Like enterprise networks, they can be a single site or multiple sites. Service provider networks usually have a device located locally to terminate the call. If they do not own the termination equipment, they transparently route the call to another ITSP device. Billing and accounting information are of utmost importance in provider networks, because their main income is generated by their telephony services. The equipment used are GWs, GKs, and softswitches. These companies usually have no reason to implement voice messaging.
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