Guest

Networking Software (IOS & NX-OS)

Cisco Unified Border Element Videoconferencing

  • Viewing Options

  • PDF (189.7 KB)
  • Feedback
Cisco Unified Border Element Videoconferencing

Table Of Contents

Cisco Unified Border Element Videoconferencing

Contents

Prerequisites for Cisco Unified Border Element Videoconferencing

Restrictions for Cisco Unified Border Element Videoconferencing

Information About Cisco Unified Border Element Videoconferencing

How to Configure Cisco Unified Border Element Videoconferencing

Configuring Audio and Video QoS Levels and Bandwidth Usage

Configuration Examples for Cisco Unified Border Element Videoconferencing

Configuring RSVP Synchronization for H.323 Slow Start

Configuring Interworking of Polycom Endpoints

Configuring SIP Video Calls with Flow-Around Media

Verifying and Monitoring Cisco Unified Border Element Videoconferencing


Cisco Unified Border Element Videoconferencing

This chapter describes how to configure the Videoconferencing for the Cisco Unified Border Element (Cisco UBE) feature. The feature provides enhanced quality of service (QoS) through RSVP synchronization with the H.323 signaling protocol and differentiated services code point (DSCP) packet marking. A Cisco Unified Border Element (in this guide also called an IP-to-IP gateway (IPIPGW)), border element (BE), or session border controller, facilitates connectivity between independent VoIP networks by enabling H.323 VoIP and videoconferencing calls from one IP network to another.


Activation Cisco Product Authorization Key (PAK)—A Product Authorization Key (PAK) is required to configure some of the features described in this guide. Before you start the configuration process, please register your products and activate your PAK at the following URL http://www.cisco.com/go/license.


Contents

Prerequisites for Cisco Unified Border Element Videoconferencing

Restrictions for Cisco Unified Border Element Videoconferencing

Information About Cisco Unified Border Element Videoconferencing

How to Configure Cisco Unified Border Element Videoconferencing

Verifying and Monitoring Cisco Unified Border Element Videoconferencing

Prerequisites for Cisco Unified Border Element Videoconferencing

Cisco Unified Border Element

Cisco IOS Release 12.2(8)T or a later release must be installed and running on your Cisco Unified Border Element.

Perform basic H.323 gateway configuration.

Perform basic H.323 gatekeeper configuration.


Note For configuration instructions, see the "Configuring H.323 Gateways" and "Configuring H.323 Gatekeepers" chapters of the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2.


Cisco Unified Border Element (Enterprise)

Cisco IOS XE Release 3.3S or a later release must be installed and running on your Cisco ASR 1000 Series Router.

Restrictions for Cisco Unified Border Element Videoconferencing

H.323-to-SIP video traffic is not supported.

H.239 for dual video (also known as Picture-in-Picture) is supported in Cisco IOS Release 12.4(20)T and later releases.

High Availability calls are will fall back to best effort on switchover. Optional or no RSVP must be configured for calls to remain active.

Video is supported with slow-start.

Dual video is not supported.

Video with fast-start and RSVP is not a supported.

Video and T.120 data are supported only with H.323 slow-start calls.

T.120 data is supported only in flow-around mode.

Video endpoints must have the same H.245 version.

Cisco Unified Border Elements that are configured for videoconferencing cannot coexist with a Multimedia Conference Manager (MCM) proxy in the same zone. See the "Migrating MCM Proxies" section for details.

Cisco Unified Border Elements are able to process audio and video calls without additional configuration.

If video calls from Cisco Unified Communications Manager directly to a Cisco UBE fail, go to the Cisco Unified Communications Manager gateway configuration and uncheck the Wait for Far End H.245 Terminal Capability Set check box.

A Cisco Unified Border Element that is configured for videoconferencing is compatible with MCM proxies. However, the following limitations apply:

The videoconferencing gateway cannot coexist with an MCM proxy in the same zone.

RSVP status depends on the type of originating and terminating gateway, as shown in the Table 1.

Information About Cisco Unified Border Element Videoconferencing

The Videoconferencing for Cisco Unified Border Element feature improves the quality, reliability, and scalability of IP videoconferencing applications. In addition to the benefits offered by the Cisco UBE feature, the videoconferencing feature provides the following functionality:

Multiple logical channels per call leg

Exchange of video and T.120 data between H.323 call legs

Exchange of H.245 miscellaneous commands and indications and generic capabilities between H.323 call legs

Far End Camera Control (FECC) support

Differentiated services code point (DSCP) marking for video streams

RSVP synchronization of H.323 calls

New vendor-specific attribute (VSA) for improved accounting of bandwidth usage

Videoconferencing for the Cisco UBE benefits include the following:

FECC enables an endpoint to control the remote camera on a video call connected through the Cisco UBE.

Cisco gateways can be configured to use the max-bit-rate VSA to report bandwidth usage to accounting servers.

Cisco Unified Border Elements can process audio and video calls without additional configuration. However, you will most likely want to set quality-of-service (QoS) levels and control how available bandwidth is divided among the calls passing through the gateway.

This section contains the following information:

MCM Proxies

QoS Levels

Bandwidth Usage

MCM Proxies

Cisco Multimedia Conference Manager (MCM) is a Cisco IOS software feature set that enables IP networks to support secure, reliable H.323 videoconferencing with advanced QoS capabilities. MCM functions as a high-performance H.323 gatekeeper and proxy, allowing network managers to control bandwidth and priority setting for H.323 videoconferencing services based on individual network configurations and capacities. These capabilities ensure appropriate allocation of network resources for videoconferencing and other critical applications running simultaneously on the network.

A Cisco Unified Border Element (Cisco UBE) that is configured for videoconferencing is compatible with MCM proxies. However, the following limitations apply:

The videoconferencing gateway cannot coexist with an MCM proxy in the same zone.

RSVP status depends on the type of originating and terminating gateway, as shown in the following Table 1.

Table 1 RSVP Status Types

Gateway Type
RSVP Status
Originating Gateway
Terminating Gateway

MCM proxy

Cisco UBE

Synchronized

Cisco UBE

MCM proxy

Not synchronized


QoS Levels

You can configure required and acceptable QoS levels on the gateway by means of the req-qos and acc-qos commands. The following levels are available:

Best-effort—Bandwidth reservation is not attempted.

Controlled-load—Synchronized RSVP is attempted. If it fails, the call is released.

Guaranteed-delay—Synchronized RSVP is attempted. If it fails, one of the following occurs:

If acceptable QoS is best effort, call setup proceeds but without bandwidth reservation.

If acceptable QoS on either gateway is anything other than best effort, the call is released.

Table 2 summarizes the results of nine call-setup scenarios based on the QoS levels configured in the dial peers at the originating and terminating gateways. It does not include cases in which the requested QoS is best effort and the acceptable QoS is something other than best effort.

Table 2 Nine Call Result Examples Based on Configured QoS Levels 

Call Scenario
Originating Gateway
Terminating Gateway
Requested QoS
Acceptable QoS
Requested QoS
Acceptable QoS
Results

1

controlled-load or guaranteed-delay

controlled-load or guaranteed-delay

controlled-load or guaranteed-delay

controlled-load or guaranteed-delay

Call proceeds only if both RSVP reservations succeed.

2

controlled-load or guaranteed-delay

controlled-load or guaranteed-delay

controlled-load or guaranteed-delay

best-effort

Call proceeds only if both RSVP reservations succeed.

3

controlled-load or guaranteed-delay

controlled-load or guaranteed-delay

best-effort

best-effort

Call is released.

4

controlled-load or guaranteed-delay

best-effort

controlled-load or guaranteed-delay

controlled-load or guaranteed-delay

Call proceeds only if both RSVP reservations succeed.

5

controlled-load or guaranteed-delay

best-effort

controlled-load or guaranteed-delay

best-effort

Call proceeds regardless of RSVP results. If RSVP reservation fails, call receives best-effort service.

6

controlled-load or guaranteed-delay

best-effort

best-effort

best-effort

Call proceeds with best-effort service.

7

best-effort

best-effort

controlled-load or guaranteed-delay

controlled-load or guaranteed-delay

Call is released.

8

best-effort

best-effort

controlled-load or guaranteed-delay

best-effort

Call proceeds with best-effort service.

9

best-effort

best-effort

best-effort

best-effort

Call proceeds with best-effort service.


Bandwidth Usage

Cisco Unified Border Elements (Cisco UBE) make bandwidth decisions based on specified or default QoS levels. The req-qos command enables you to specify how much bandwidth is used by individual calls passing through the Cisco UBE. You can specify default and maximum amounts of bandwidth to be requested for each call. Bandwidth usage varies depending on if the gateway is the originating gateway or terminating gateway.

Originating Cisco Unified Border Element

If you set the required QoS level as the default for audio (by means of the req-qos guaranteed-delay audio bandwidth default command and keywords), an audio reservation is made for the default value of 64 kbps.

Normally, a video RSVP reservation is made using the value in the SETUP message bearer capability information element (IE). If this value is zero (such as with Microsoft NetMeeting), the value specified with the video bandwidth default keyword is used.

When you configure audio streams for either controlled-load or guaranteed-delay and configure maximum values for both audio and video, the setup is rejected if the value from the bearer-capability IE exceeds the sum total of the audio bandwidth max and video bandwidth max commands. The maximum values are also checked at the time the audio and video media channels are opened. The Cisco UBE never reserves more bandwidth than the values specified with the max keyword.


Note If you do not set a maximum for either audio or video, the bearer-capability IE is not checked against maximum values during SETUP.


Terminating Cisco Unified Border Element

The value in the bearer-capability IE is not considered in the badwidth decision. Instead, the audio and video bandwidth values from the SETUP message nonstandard field are used. These values are compared with the maximum values for audio and video max configured on the terminating Cisco UBE. The smaller of the two values is used for RSVP.

Table 3 summarizes the call-setup scenarios based on the configured RSVP behavior in the dial peers at the originating and terminating gateways.

Table 3 Call Results Based on RSVP Behavior 

Sync Mode
RSVP Mode
RSVP Result
Behavior

Sync

Requested, not best effort

Audio and video RSVP failed.

Do nothing.

Acceptable, not best effort

Requested, not best effort

Audio RSVP failed.

End the call.

Acceptable, best effort

Video RSVP failed.

End the call.

Nonsync

Requested, not best effort

Audio and video RSVP failed.

Do nothing.

Acceptable, best effort

Requested, not best effort

Audio RSVP failed.

End the call.

Acceptable, not best effort

Video RSVP failed.

Close the video channel.


How to Configure Cisco Unified Border Element Videoconferencing

Migrating MCM Proxies

Configuring Via-Zone Gatekeepers for Video Calls

Configuring Audio and Video QoS Levels and Bandwidth Usage

Configuring RSVP Synchronization for H.323 Slow Start

Configuring Interworking of Polycom Endpoints

Configuring SIP Video Calls with Flow-Around Media

Verifying and Monitoring Cisco Unified Border Element Videoconferencing

Migrating MCM Proxies

Converting MCM Zones

A network that uses MCM usually consists of multiple zones, each of which includes at least one gatekeeper and one MCM proxy.

Migrate a network from MCM proxies to videoconferencing gateways on a zone-by-zone basis. When a zone is converted, replace all of the MCM proxies in that zone with Cisco Unified Border Element videoconferencing gateways.

Converting Individual Devices

Frequently the gatekeeper and the MCM proxy are collocated on the same router. The videoconferencing gateway cannot reside on the same device with the gatekeeper, so you need an additional router to perform videoconferencing gateway functions.

You can reuse the router that hosted the collocated gatekeeper and MCM proxy for the via-zone gatekeeper. Upgrade to a Cisco IOS release that supports via-zones. Reuse the original gatekeeper-configuration data during configuration of the new via-zone gatekeeper as appropriate. Remove the portions related to the MCM proxy and replace them with the equivalent via-zone configuration.


Note If a local zone is configured for via-zone, the Cisco UBE is used for all calls.


Configuring Via-Zone Gatekeepers for Video Calls

To configure via-zone gatekeepers for video calls, perform the steps in this section.


Note Video calls can take advantage of the benefits offered by via-zone gatekeeper processing. For more information, see the "Configuring Via-Zones" section of the Cisco Unified Border Element with Gatekeeper guide.


Restrictions for Configuring Via-Zone Gatekeepers for Video Calls

Although gatekeepers can support multiple local zones, call routing between a local zone and a via zone on the same gatekeeper is not supported in Cisco IOS Release 12.2(4)T and earlier releases. Via-zone gatekeepers must be dedicated to their own via-zones.

SUMMARY STEPS

1. enable

2. configure terminal

3. gatekeeper

4. zone local gatekeeper-name domain-name [ras-IP-address] [invia inbound-gatekeeper | outvia outbound gatekeeper [enable-intrazone]]

5. exit

DETAILED STEPS
 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enables privileged EXEC mode.

Enter your password if prompted.

Step 2 

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

gatekeeper

Example:

Router(config)# gatekeeper

Enters gatekeeper configuration mode.

Step 4 

zone local gatekeeper-name domain-name [ras-IP-address] [invia inbound-gatekeeper | outvia outbound gatekeeper [enable-intrazone]]

Example:

Router(config-gk)# zone local termGK example.com 10.16.193.158 invia gatekeeper1 outvia gatekeeper2 enable-intrazone

Defines the local gatekeeper zone.

Keywords and arguments are as follows:

gatekeeper-name—Gatekeeper name or zone name.

domain-name—Domain name served by this gatekeeper.

ras-IP-address—IP address of one of the interfaces on the gatekeeper.

invia inbound-gatekeeper—Name of the gatekeeper for calls entering this zone.

outvia outbound-gatekeeper—Name of the gatekeeper for calls leaving this zone.

enable-intrazone—All intrazone calls are forced to use the via-zone gatekeeper.

Note You can specify invia and outvia gatekeepers to be used for intrazone video calls. You can also specify enable-intrazone to force all intrazone calls to use the via-zone gatekeeper.

Step 5 

exit

Example:

Router(config-gk)# exit

Exits the current mode.

Configuring Audio and Video QoS Levels and Bandwidth Usage

To configure Audio and Video QoS Levels and Bandwidth Usage, perform the steps in this section.


Note The following steps include sample settings that may not be appropriate for your network.


SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. acc-qos guaranteed-delay audio

5. acc-qos guaranteed-delay video

6. req-qos guaranteed-delay audio bandwidth default [value] max [value]

7. req-qos guaranteed-delay video bandwidth default [value] max [value]

8. ip qos dscp video

9. exit

DETAILED STEPS
 
Command
Purpose

Step 1 

enable

Example:

Router> enable

Enters privileged EXEC mode.

Enter your password when prompted.

Step 2 

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

dial-peer voice tag voip

Example:

Router(config)# dial-peer voice tag voip

Enters dial peer configuration mode for the specified VoIP dial peer.

Step 4 

acc-qos guaranteed-delay audio

Example:

Router(config-dial-peer)# acc-qos guaranteed-delay audio

Sets an acceptable QoS for audio traffic. RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.

Note You cannot use the acc-qos command by itself. You must also use req-qos to specify a desired QoS for audio traffic. See Step 6.

Step 5 

acc-qos guaranteed-delay video

Example:

Router(config-dial-peer)# acc-qos guaranteed-delay video

Sets an acceptable QoS for video traffic. RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.

Note You cannot use the acc-qos command by itself. You must also use req-qos to specify a desired QoS for video traffic. See Step 7.

Step 6 

req-qos guaranteed-delay audio bandwidth default [kilobits] max [kilobits]

Example:
Router(config-dial-peer)# req-qos 
guaranteed-delay audio bandwidth default 15 max 
45

Sets required QoS for audio traffic. RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded. Keywords and arguments are as follows:

default [kilobits]—Default audio bandwidth for RSVP, in kbps. Range: 1 to 64. Default: 64.

max [kilobits]—Maximum audio bandwidth for RSVP, in kbps. Range: 1 to 64. Default: no maximum.

Step 7 

req-qos guaranteed-delay video bandwidth default [kilobits] max [kilobits]

Example:
Router(config-dial-peer)# req-qos 
guaranteed-delay video bandwidth default 12 max 
65

Sets required QoS for video traffic. RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded. Keywords and arguments are as follows:

default [kilobits]—Default video bandwidth for RSVP, in kbps. Range: 1 to 5000. Default: 384.

max [kilobits]—Maximum video bandwidth for RSVP, in kbps. Range: 1 to 5000. Default: no maximum.

Step 8 

ip qos dscp [value] video [rsvp-none | rsvp-pass | rsvp-fail]

Example:

Router(config-dial-peer)# ip qos dscp 65 video rsvp-none

Sets the DSCP for QoS. In this case, the command allows DSCP marking of RTP packets for the video stream. Keywords and arguments are as follows:

value—DSCP value. Range: 0 to 63.

video rsvp-none—Applies DSCP to a video stream with no RSVP reservations.

video rsvp-pass—Applies DSCP to avideo stream with successful RSVP reservations.

video rsvp-fail—Applies DSCP to avideo stream with failed RSVP reservations.

Step 9 

exit

Example:

Router(config-dial-peer)# exit

Exits the current mode.

Configuration Examples for Cisco Unified Border Element Videoconferencing

QoS for Audio and Video on One Gateway: Example

QoS for Audio and Video on Two Gateways: Example

QoS for Audio and Video on One Gateway: Example

The following example shows QoS for audio and video configured on a Cisco Unified Border Element. Note that this example uses values and settings that may not be appropriate for your network.

!
voice service voip
 no allow-connections any to pots
 no allow-connections pots to any
 allow-connections h323 to h323
 h323
  no call sync-rsvp slow-start
!
!
voice class h323 1
  no call sync-rsvp slow-start
!
!
interface FastEthernet0/0
 ip address 10.1.1.2 255.255.255.0
 ip route-cache same-interface
 h323-gateway voip interface
 h323-gateway voip id zone1-gk ipaddr 10.1.1.1 1718
 h323-gateway voip tech-prefix 1#
 h323_gateway voip bind srcaddr 10.1.1.2
 ip rsvp bandwidth 7000 1000
!
!
dial-peer voice 100 voip
 voice-class h323 1
 req-qos guaranteed-delay audio bandwidth default 16 max 32
 req-qos guaranteed-delay video bandwidth default 320 max 768
 acc-qos guaranteed-delay audio
 acc-qos guaranteed-delay video
 ip qos dscp af11 media
 ip qos dscp af21 signaling
 ip qos dscp af33 video rsvp-none
 ip qos dscp af31 video rsvp-pass
 ip qos dscp af32 video rsvp-fail
 codec transparent

QoS for Audio and Video on Two Gateways: Example

The following example shows the dial peers for two Cisco Unified Border Elements that exchange video calls. Each gateway is connected to an endpoint that does not support RSVP; however, RSVP is used between the Cisco UBEs. One endpoint has an E.164 address of 1231000, and the other endpoint has an E.164 address of 4569000. Because the endpoints do not support RSVP, the gateways must have two dial peers for each call leg, one that prevents RSVP reservations to the endpoints and one that allows RSVP between the gateways.

Cisco Unified Border Element Connected to 1231000

dial-peer voice 123 voip
 description dial-peer incoming from ip-ip gateway
 incoming called-number 123....
 session target ras
 req-qos guaranteed-delay audio
 req-qos guaranteed-delay video
 acc-qos guaranteed-delay audio
 acc-qos guaranteed-delay video
 codec transparent
!
dial-peer voice 456 voip
 description dial-peer incoming from video endpoint
 incoming called-number 456....
 session target ras
 codec transparent
!
dial-peer voice 4569 voip
 description dial-peer outgoing to ip-ip gateway
 destination-pattern 456....
 session target ras
 req-qos guaranteed-delay audio
 req-qos guaranteed-delay video
 acc-qos guaranteed-delay audio
 acc-qos guaranteed-delay video
 codec transparent
!
dial-peer voice 1231 voip
 description dial-peer outgoing to video endpoint
 destination-pattern 123....
 session target ras
 codec transparent
!

Cisco Unified Border Element Connected to 4569000

dial-peer voice 123 voip
 description dial-peer incoming from video endpoint
 incoming called-number 123....
 session target ras
 codec transparent
!
dial-peer voice 456 voip
 description dial-peer incoming from ip-ip gateway
 incoming called-number 456....
 session target ras
 req-qos guaranteed-delay audio
 req-qos guaranteed-delay video
 acc-qos guaranteed-delay audio
 acc-qos guaranteed-delay video
 codec transparent
!
dial-peer voice 1231 voip
 description dial-peer outgoing to ip-ip gateway
 destination-pattern 123....
 session target ras
 req-qos guaranteed-delay audio
 req-qos guaranteed-delay video
 acc-qos guaranteed-delay audio
 acc-qos guaranteed-delay video
 codec transparent
!
dial-peer voice 4569 voip
 description dial-peer outgoing to video endpoint
 destination-pattern 456....
 session target ras
 codec transparent
!

Configuring RSVP Synchronization for H.323 Slow Start

To configure RSVP synchronization for H.323 slow start for all H.323 calls, perform the steps in this section.


Note This task is optional; RSVP synchronization is enabled by default.


SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip

4. h323

5. call start {fast | slow | system}[sync-rsvp slow-start]

6. exit

DETAILED STEPS
 
Command
Purpose

Step 1 

enable

Example:

Router> enable

Enters privileged EXEC mode.

Enter your password when prompted.

Step 2 

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

voice service voip

Example:

Router(config)# voice service voip

Enters voice service configuration mode.

Step 4 

h323

Example:

Router(conf-voi-serv)# h323

Enters H.323 configuration mode.

Step 5 

call start {fast | slow | system}[sync-rsvp 
slow-start]
Example:

Router(config-serv-h323)# call start slow sync-rsvp slow-start

Forces an H.323 gateway to use fast-connect or slow-connect procedures for a dial peer. Use the sync-rsvp slow-start keyword to enable RSVP synchronization for slow-start calls. Keywords are as follows:

fast—Fast-connect procedures

slow—Slow-connect procedures

system—Voice-service configuration

sync-rsvp slow-start—RSVP synchronization for slow-start calls

Default: system

Step 6 

exit

Example:

Router(config-class)# exit

Exits the current mode.

Configuring Interworking of Polycom Endpoints

To configure interworking between Polycom endpoints, perform the steps in this section.

Restrictions for Configuring Interworking of Polycom Endpoints

Interworking between Polycom endpoints is determined by the software version running on each endpoint.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip

4. h323

5. h225 h225 id-passthru

6. exit

DETAILED STEPS
 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enables privileged EXEC mode. Enter your password if prompted.

Step 2 

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

voice service voip

Example:

Router(config)# voice service voip

Enters voice service configuration mode.

Step 4 

h323

Example:

Router(config-voice-serv)# h323

Enters H.323 voice service configuration mode.

Step 5 

h225 id-passthru

Example:

Router(config-serv-h323)# h225 id-passthru

Enables signaling between video endpoints with different H.225 versions.

Step 6 

exit

Example:

Router(config-serv-h323)# exit

Exits the current mode.

Configuring SIP Video Calls with Flow-Around Media

The SIP Video Calls with Flow-Around Media allows user to configure SIP video calls to be placed on the Cisco Unified Border Element (Cisco UBE) where the media flows around the Cisco UBE from endpoint to endpoint. The default behavior is to have SIP video calls pass directly from endpoint to endpoint,

Restrictions for Configuring SIP Video Calls with Flow-Around Media

SIP video calls with flow-around media is supported in Cisco IOS Release 12.4(20)T and later.

SIP video calls with flow-through media is supported in Cisco IOS Release 12.4(15)XZ and earlier.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip

4. h323

5. media [flow-around | flow-through | forking | monitoring [max-calls] | statistics | transcoder high-density | anti-trombone | sync-streams]

6. exit

DETAILED STEPS
 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enables privileged EXEC mode.

Enter your password if prompted.

Step 2 

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

voice service voip

Example:

Router(config)# voice service voip

Enters voice service configuration mode.

Step 4 

h323

Example:

Router(conf-voi-serv)# h323

Enters H.323 voice-service configuration mode.

Step 5 

media [flow-around | flow-through | forking | monitoring [max-calls] | statistics | transcoder high-density | anti-trombone | sync-streams]

Example:

Router(conf-voi-serv)#

Passes media packets directly between the endpoints, without the intervention of the Cisco UBE and enables signaling services.

Step 6 

exit

Example:

Router(config-serv-h323)# exit

Exits the current mode.

Verifying and Monitoring Cisco Unified Border Element Videoconferencing

To verify, monitor, and maintain audio and video calls, perform the following steps (listed alphabetically).

SUMMARY STEPS

1. show call active video

2. show call history video

3. show dial-peer voice

4. show ip rsvp reservation

5. show running-config

6. show sccp connections

DETAILED STEPS


Step 1 show call active video

Use this command to display call statistics, including video bytes and packets received, video bytes and packets transmitted, bandwidth used, and UDP ports used, for active calls.

Step 2 show call history video

Use this command to display the same call statistics for all calls.

Step 3 show dial-peer voice

Use this command to display dial-peer statistics, including default and maximum bandwidth values for audio and video and DSCP marking for video.

Step 4 show ip rsvp reservation

Use this command to display RSVP-related receiver information currently in the database.

Step 5 show running-config

Use this command to verify audio and video QoS.

!
interface FastEthernet0/0
 ip address 10.1.1.5 255.255.255.0
 ip route-cache same-interface
 h323-gateway voip interface
 h323-gateway voip id zone1-gk ipaddr 10.1.1.1 1718
 h323-gateway voip tech-prefix 1#
 h323_gateway voip bind srcaddr 10.1.1.5
 ip rsvp bandwidth 7000 1000
!
!
dial-peer voice 100 voip
 voice-class h323 1
 req-qos guaranteed-delay audio bandwidth default 16 max 32
 req-qos guaranteed-delay video bandwidth default 320 max 768
 acc-qos guaranteed-delay audio
 acc-qos guaranteed-delay video
 ip qos dscp af11 media
 ip qos dscp af21 signaling
 ip qos dscp af33 video rsvp-none
 ip qos dscp af31 video rsvp-pass
 ip qos dscp af32 video rsvp-fail
 codec transparent
!

Step 6 show sccp connections

Use this command to display Skinny Client Control Protocol (SCCP) connection details.