Cisco Hoot and Holler over IP

This feature module describes the Cisco Hoot and Holler over IP feature and contains the following sections:

Hoot and Holler networks came into being more than 40 years ago when local concentrations of small specialized businesses with common, time-critical informational interests—junkyards, for example—began to install their own phone wires, speakers (called "squawk boxes"), and microphones between their businesses to ask each other about parts customers needed. This network functioned as a crude, do-it-yourself, businesses-to-businesses intercom system.

Hoot and Holler broadcast audio network systems have since evolved into the specialized leased-line networks used by financial and brokerage firms to trade stocks and currency futures and the accompanying time-critical information such as market updates and the morning reports.

Now, users of various forms of Hoot and Holler networks include brokerages, news agencies, publishers, weather bureaus, transportation providers, power plant operators, manufacturers, collectibles dealers, talent agencies, and nationwide salvage yard organizations.

The voice multicasting feature on Cisco 2600 and Cisco 3600 series routers uses Cisco Voice over IP (VoIP) technology to create a permanently connected point-to-multipoint Hoot and Holler network over an IP connection.

Voice multicasting telephones can be connected to network routers in the following ways:

Connect a four-wire E&M telephone, which has no dial and is always off-hook, directly to an E&M voice interface card that is installed in a voice network module. Configure the E&M interface for four-wire trunk operation. For information about configuring E&M interfaces, see the
Cisco IOS Multiservice Applications Configuration Guide, Release 12.1.
Connect a conventional telephone to a PBX that is connected to an E&M voice interface card.
Connect a conventional telephone to a PBX that is connected through a T1 line to a multiflex trunk interface card that is installed in a high-density voice network module.

Note The voice multicasting feature supports only one T1 line per high-density voice network module.

Benefits

Cisco Hoot and Holler over IP:

Eliminates yearly reoccurring circuit-switched telecom charges (toll-bypass).
Eliminates the need for leased-lines and the accompanying charges.
Enables idle bandwidth to be reclaimed by data applications.
Reduces the number of Hoot and Holler bridges required.
Improves Hoot and Holler network manageability.
Reduces the time to troubleshoot a problem from hours to minutes.
Reduces the time to provision bandwidth from days to a few hours.
Increases productivity through future applications (such as IP/TV, and IP turret support)

Restrictions

IP/TV does not support mixing of audio streams.
IP/TV supports only G.711 codec (64 kbps bandwidth).
SDR support will be limited to IP/TV Viewer and Content Manager as they are used today.

Supported Platforms

Cisco 2600 series
Cisco 3600 series

Supported Standards, MIBs, and RFCs

Standards

No new or modified standards are supported by this feature.

MIBs

No new or modified MIBs are supported by this feature.

For descriptions of supported MIBs and how to use MIBs, see the Cisco MIB web site on CCO at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.

RFCs

No new or modified RFCs are supported by this feature.

Configuration Tasks

See the following sections for configuring Multicast Hoot and Holler over IP:

Configuring Voice Ports (Required)

Command Purpose

Step 1

Router(config)# ip multicast-routing
Enables multicast routing.

Step 2

Router(config)# voice class permanent tag1

Defines voice class for transmit-receive mode.

Step 3

Router(config-class)# signal timing oos timeout
disabled

Disables signaling loss detection.

Step 4

Router(config-class)# signal keepalive number

Specifies keepalive signaling packet interval.

Step 5

Router(config-class)# voice class permanent tag2
Defines voice class for receive-only mode.

Step 6

Router(config-class)# signal pattern oos receive 0000
Specifies the received signal pattern.

Step 7

Router(config-class)# signal timing oos suppress-all
seconds
If the transmit out-of-service pattern (from the PBX to the network) matches for the time specified, the router stops sending packets to the network.

Step 8

Router(config-class)# signal keepalive number

Specifies keepalive signaling packet interval.

Step 9

Router(config)# voice-port
router-slot/voice-slot/VIC-port
Selects the voice port to configure.

Step 10

Router(config-voice-port)# voice-class permanent tag1
Uses voice class tag1 for the port that is allowed to speak.

Step 11

Router(config-voice-port)# vad
Enables voice activity detection (VAD). This is the default setting and should not be changed.

Step 12

Router(config-voice-port)# connection trunk
phone-number
Ties the voice port to a phone number.

Step 13

Router(config-voice-port)# music-threshold threshold
Sets the music threshold to make VAD less sensitive.

Step 14

Router(config-voice-port)# operation 4-wire
Specifies 4-wire operation.

Step 15

Router(config-voice-port)# voice-port
router-slot/voice-slot/VIC-port
Selects another voice port.

Step 16

Router(config-voice-port)# voice-class permanent tag2
Uses voice class tag2 for the receive-only port.

Step 17

Router(config-voice-port)# vad
Enables VAD.

Step 18

Router(config-voice-port)# connection trunk
phone-number
Ties the voice port to the same phone number as in Step 12.

Step 19

Router(config-voice-port)# music-threshold threshold
Sets the music threshold to make VAD less sensitive.

Step 20

Router(config-voice-port)# operation 4-wire
Specifies 4-wire operation.

Step 21

Router(config)# type { 1 | 2 | 3 | 5 }
Select the appropriate E&M interface type.

Type 1 indicates the following lead configuration (default—this is the recommended option):

E—output, relay to ground

M—input, referenced to ground

Type 2 indicates the following lead configuration:

E—output, relay to SG

M—input, referenced to ground

SB—feed for M, connected to -48V

SG—return for E, galvanically isolated from ground

Type 3 indicates the following lead configuration:

E—output, relay to ground

M—input, referenced to ground

SB—connected to -48V

SG—connected to ground

Type 5 indicates the following lead configuration:

E—output, relay to ground

M—input, referenced to -48V.

Step 22

Router(config)# signal { wink-start | immediate | delay-dial }
Configure the signalling type for E&M voice ports. The default is wink-start. Select immediate for Hoot and Holler applications.

	Command	Purpose
Step 1	Router(config)# ip multicast-routing	Enables multicast routing.
Step 2	Router(config)# voice class permanent tag1	Defines voice class for transmit-receive mode.
Step 3	Router(config-class)# signal timing oos timeout disabled	Disables signaling loss detection.
Step 4	Router(config-class)# signal keepalive number	Specifies keepalive signaling packet interval.
Step 5	Router(config-class)# voice class permanent tag2	Defines voice class for receive-only mode.
Step 6	Router(config-class)# signal pattern oos receive 0000	Specifies the received signal pattern.
Step 7	Router(config-class)# signal timing oos suppress-all seconds	If the transmit out-of-service pattern (from the PBX to the network) matches for the time specified, the router stops sending packets to the network.
Step 8	Router(config-class)# signal keepalive number	Specifies keepalive signaling packet interval.
Step 9	Router(config)# voice-port router-slot/voice-slot/VIC-port	Selects the voice port to configure.
Step 10	Router(config-voice-port)# voice-class permanent tag1	Uses voice class tag1 for the port that is allowed to speak.
Step 11	Router(config-voice-port)# vad	Enables voice activity detection (VAD). This is the default setting and should not be changed.
Step 12	Router(config-voice-port)# connection trunk phone-number	Ties the voice port to a phone number.
Step 13	Router(config-voice-port)# music-threshold threshold	Sets the music threshold to make VAD less sensitive.
Step 14	Router(config-voice-port)# operation 4-wire	Specifies 4-wire operation.
Step 15	Router(config-voice-port)# voice-port router-slot/voice-slot/VIC-port	Selects another voice port.
Step 16	Router(config-voice-port)# voice-class permanent tag2	Uses voice class tag2 for the receive-only port.
Step 17	Router(config-voice-port)# vad	Enables VAD.
Step 18	Router(config-voice-port)# connection trunk phone-number	Ties the voice port to the same phone number as in Step 12.
Step 19	Router(config-voice-port)# music-threshold threshold	Sets the music threshold to make VAD less sensitive.
Step 20	Router(config-voice-port)# operation 4-wire	Specifies 4-wire operation.
Step 21	Router(config)# type { 1 \| 2 \| 3 \| 5 }	Select the appropriate E&M interface type. Type 1 indicates the following lead configuration (default—this is the recommended option): E—output, relay to ground M—input, referenced to ground Type 2 indicates the following lead configuration: E—output, relay to SG M—input, referenced to ground SB—feed for M, connected to -48V SG—return for E, galvanically isolated from ground Type 3 indicates the following lead configuration: E—output, relay to ground M—input, referenced to ground SB—connected to -48V SG—connected to ground Type 5 indicates the following lead configuration: E—output, relay to ground M—input, referenced to -48V.
Step 22	Router(config)# signal { wink-start \| immediate \| delay-dial }	Configure the signalling type for E&M voice ports. The default is wink-start. Select immediate for Hoot and Holler applications.

Configuring the Virtual Interface (Required)

Command Purpose

Step 1

Router(config)# interface virtual-interface
Defines a virtual interface for multicast fast switching. Routers joining the same session must have their virtual interfaces on different subnets. Otherwise packets are not switched to the IP network.

Step 2

Router(config-if)# ip address address subnet-mask
Assigns the IP address and subnet mask for the virtual interface.

Step 3

Router(config-if)# ip pim { sparse mode | dense-mode | sparse-dense-mode }
Specifies Protocol Independent Multicast (PIM). Whatever mode you choose should match all the interfaces in all the routers of your network.

Configure the virtual interface for multicast fast switching.

	Command	Purpose
Step 1	Router(config)# interface virtual-interface	Defines a virtual interface for multicast fast switching. Routers joining the same session must have their virtual interfaces on different subnets. Otherwise packets are not switched to the IP network.
Step 2	Router(config-if)# ip address address subnet-mask	Assigns the IP address and subnet mask for the virtual interface.
Step 3	Router(config-if)# ip pim `{` sparse mode `\|` dense-mode `\|` sparse-dense-mode `}`	Specifies Protocol Independent Multicast (PIM). Whatever mode you choose should match all the interfaces in all the routers of your network.

Configuring Voice Ports in High-Density Voice Network Modules (Required)

A multiflex trunk interface card in a high-density voice network module requires special voice-port configuration.

Command Purpose

Step 1

Router(config)# voice-card number

Selects the card to configure.

Step 2

Router(config-voicecard)# codec complexity high
Codec complexity must be high. Voice multicasting does not support medium complexity, which is the default.

Step 3

Router(config)# controller t1 slot/port
Selects the T1 controller to configure.

Step 4

Router(config-controller)# ds0-group ds0-group-number
timeslots timeslot-list type e&m-immediate-start
Maps each DS0 group to a timeslot with the same number. This command is repeated for each group from 1 to 23.

Step 5

Router(config)# voice-port slot/port:ds0-group-number
Maps each DS0 to voice port slot/port:ds0-group-number. This command is repeated for each group number from 1 to 23.

Step 6

Router(config-voice-port)# connection trunk phone-number
Ties the connection trunk to a phone number. This command is repeated for each DS0 group. All groups use the same phone number.

	Command	Purpose
Step 1	Router(config)# voice-card number	Selects the card to configure.
Step 2	Router(config-voicecard)# codec complexity high	Codec complexity must be high. Voice multicasting does not support medium complexity, which is the default.
Step 3	Router(config)# controller t1 slot/port	Selects the T1 controller to configure.
Step 4	Router(config-controller)# ds0-group ds0-group-number timeslots timeslot-list type e&m-immediate-start	Maps each DS0 group to a timeslot with the same number. This command is repeated for each group from 1 to 23.
Step 5	Router(config)# voice-port slot/port:ds0-group-number	Maps each DS0 to voice port slot/port:ds0-group-number. This command is repeated for each group number from 1 to 23.
Step 6	Router(config-voice-port)# connection trunk phone-number	Ties the connection trunk to a phone number. This command is repeated for each DS0 group. All groups use the same phone number.

Configuring Dial Peers (Required)

Command Purpose

Step 1

Router(config)# dial-peer voice tag voip
Assigns a tag to the VOIP dial peer.

Step 2

Router(config-dial-peer)# destination-pattern phone-number
The destination pattern for the VOIP dial peer must match the connection trunk string for the corresponding voice port.

Step 3

Router(config-dial-peer)# session protocol multicast
Enables multicasting. This step is mandatory for voice multicasting.

Step 4

Router(config-dial-peer)# session target ipv4:address:port
Assigns the session target for voice multicasting dial peers. This is a multicast address in the range 224.0.1.0 to 239.255.255.255, and must be the same for all ports in a session.

The audio RTP port is an even number in the range 16384 to 32767, and must also be the same for all ports in a session.

Step 5

Router(config-dial-peer)# ip precedence number
Specifies the IP precedence.

Step 6

Router(config-dial-peer)# codec {g711alaw | g711ulaw |
g726r32 | g729ar8 | g729r8}
Configures the codec. You must configure the same codec on all dial peers in a session.

Only G.711, G.726, and G.729 codecs are supported. When the default codec, G.729, is used, it does not appear in the configuration.

	Command	Purpose
Step 1	Router(config)# dial-peer voice tag voip	Assigns a tag to the VOIP dial peer.
Step 2	Router(config-dial-peer)# destination-pattern phone-number	The destination pattern for the VOIP dial peer must match the connection trunk string for the corresponding voice port.
Step 3	Router(config-dial-peer)# session protocol multicast	Enables multicasting. This step is mandatory for voice multicasting.
Step 4	Router(config-dial-peer)# session target ipv4:address:port	Assigns the session target for voice multicasting dial peers. This is a multicast address in the range 224.0.1.0 to 239.255.255.255, and must be the same for all ports in a session. The audio RTP port is an even number in the range 16384 to 32767, and must also be the same for all ports in a session.
Step 5	Router(config-dial-peer)# ip precedence number	Specifies the IP precedence.
Step 6	Router(config-dial-peer)# codec {g711alaw \| g711ulaw \| g726r32 \| g729ar8 \| g729r8}	Configures the codec. You must configure the same codec on all dial peers in a session. Only G.711, G.726, and G.729 codecs are supported. When the default codec, G.729, is used, it does not appear in the configuration.

Configuring Ethernet (Optional)

Command Purpose

Step 1

Router(config)# interface ethernet slot/port
Configures the physical interface for transmitting multicast packets.

Step 2

Router(config-if)# ip address address subnet-mask
Assigns the IP address and subnet mask for the interface.

Step 3

Router(config-if)# ip pim { sparse mode | dense-mode | sparse-dense-mode }
Specifies Protocol Independent Multicast (PIM). Whatever mode you choose should match all the interfaces in all the routers of your network.

Step 4

Router(config-if)# ip sap listen
Listens to packets of Session Announcement Protocol.

Step 5

Router(config-if)# no shutdown
Enables the interface.

	Command	Purpose
Step 1	Router(config)# interface ethernet slot/port	Configures the physical interface for transmitting multicast packets.
Step 2	Router(config-if)# ip address address subnet-mask	Assigns the IP address and subnet mask for the interface.
Step 3	Router(config-if)# ip pim `{` sparse mode `\|` dense-mode `\|` sparse-dense-mode `}`	Specifies Protocol Independent Multicast (PIM). Whatever mode you choose should match all the interfaces in all the routers of your network.
Step 4	Router(config-if)# ip sap listen	Listens to packets of Session Announcement Protocol.
Step 5	Router(config-if)# no shutdown	Enables the interface.

Configuring Quality of Service (Recommended)

Voice traffic is much more sensitive to timing variations than data traffic. For good voice performance over a WAN, you might need to configure your data network so voice packets are not lost or delayed. This section shows how to improve quality of service (QoS) for voice multicasting over a Frame Relay serial connection.

Command Purpose

Step 1

Router(config)# interface serial slot/port
Specifies the interface to configure.

Step 2

Router(config-if)# encapsulation frame-relay
Configures Frame Relay encapsulation.

Step 3

Router(config-if)# frame-relay traffic-shaping
Configures Frame Relay traffic shaping.

Step 4

Router(config-if)# no frame-relay broadcast-queue
Disables the broadcast queue.

Step 5

Router(config-if)# interface serial
slot/port.subinterface point-to-point
Specifies the subinterface to configure.

Step 6

Router(config-if)# ip address subnet-mask
Assigns an IP address and subnet mask.

Step 7

Router(config-if)# ip pim sparse-dense-mode
Configures PIM sparse-dense mode.

Step 8

Router(config-if)# frame-relay class name
Specifies the Frame Relay map class to associate with this subinterface.

Step 9

Router(config-if)# frame-relay interface-dlci number
Assigns a DLCI to the interface.

Step 10

Router(config-if)# frame-relay ip rtp header-compression
Enables IP RTP header compression.

Step 11

Router(config-if)# map-class frame-relay name
Creates the map class to be associated with the subinterface.

Step 12

Router(config-map-class)# frame-relay cir bps
Specifies the committed information rate (CIR).

Step 13

Router(config-map-class)# frame-relay bc bits
Specifies the committed burst size.

Step 14

Router(config-map-class)# frame-relay mincir bps
Specifies the minimum acceptable CIR.

Step 15

Router(config-map-class)# no frame-relay adaptive-shaping
Disables adaptive traffic shaping.

Step 16

Router(config-map-class)# frame-relay fair-queue
Enables weighted fair queueing.

Step 17

Router(config-map-class)# frame-relay fragment
fragment_size
Enables fragmentation of Frame Relay frames.

Step 18

Router(config-map-class)# frame-relay ip rtp priority
audio-port number-of-ports bandwidth
The first number is the audio port. The second number is the number of consecutive audio ports to which the IP RTP priority queuing applies. The third number is the bandwidth, which should equal the bandwidth needed for each call multiplied by the number of calls.

	Command	Purpose
Step 1	Router(config)# interface serial slot/port	Specifies the interface to configure.
Step 2	Router(config-if)# encapsulation frame-relay	Configures Frame Relay encapsulation.
Step 3	Router(config-if)# frame-relay traffic-shaping	Configures Frame Relay traffic shaping.
Step 4	Router(config-if)# no frame-relay broadcast-queue	Disables the broadcast queue.
Step 5	Router(config-if)# interface serial slot/port.subinterface point-to-point	Specifies the subinterface to configure.
Step 6	Router(config-if)# ip address subnet-mask	Assigns an IP address and subnet mask.
Step 7	Router(config-if)# ip pim sparse-dense-mode	Configures PIM sparse-dense mode.
Step 8	Router(config-if)# frame-relay class name	Specifies the Frame Relay map class to associate with this subinterface.
Step 9	Router(config-if)# frame-relay interface-dlci number	Assigns a DLCI to the interface.
Step 10	Router(config-if)# frame-relay ip rtp header-compression	Enables IP RTP header compression.
Step 11	Router(config-if)# map-class frame-relay name	Creates the map class to be associated with the subinterface.
Step 12	Router(config-map-class)# frame-relay cir bps	Specifies the committed information rate (CIR).
Step 13	Router(config-map-class)# frame-relay bc bits	Specifies the committed burst size.
Step 14	Router(config-map-class)# frame-relay mincir bps	Specifies the minimum acceptable CIR.
Step 15	Router(config-map-class)# no frame-relay adaptive-shaping	Disables adaptive traffic shaping.
Step 16	Router(config-map-class)# frame-relay fair-queue	Enables weighted fair queueing.
Step 17	Router(config-map-class)# frame-relay fragment fragment_size	Enables fragmentation of Frame Relay frames.
Step 18	Router(config-map-class)# frame-relay ip rtp priority audio-port number-of-ports bandwidth	The first number is the audio port. The second number is the number of consecutive audio ports to which the IP RTP priority queuing applies. The third number is the bandwidth, which should equal the bandwidth needed for each call multiplied by the number of calls.

Configuration Examples

This section provides a series of configuration examples that help you to become familiar with voice multicasting. These examples also show how to ensure that each configuration is working properly before proceeding to the next step.

Voice Multicasting over an Ethernet LAN Example

Figure 1 shows the simplest configuration. Two routers are connected to each other over an Ethernet LAN. One E&M phone is connected to each router.

Figure 1: Voice Multicasting over a LAN (One Session)

Interface Vif1

Note that in all of the configuration examples the routers are configured with an "interface Vif1." This is a virtual interface that is similar to a loopback interface—it is a logical IP interface that is always up when the router is active. In addition, it must be configured so that the Cisco Hoot and Holler over IP packets that are locally mixed on the DSPs can be fast-switched along with the other data packets. This interface needs to reside on its own unique subnet, and that subnet should be included in the routing protocol updates (RIP, OSPF, etc.).

Voice Port Configuration

In router "Abbott", the phone is connected to voice port 2/0/0, using the router-slot/voice-slot/VIC-port numbering convention. This voice port is configured as in the following example:

hostname abbott
!Enable multicast routing
!
ip multicast-routing
!
!Define voice class for transmit-receive mode with tag 1
!Disable signaling loss detection
!Send keepalive packet every 65 seconds
!
voice class permanent 1
signal timing oos timeout disabled
signal keepalive 65535
!
!Define voice class for receive-only mode with tag 2.
!
voice class permanent 2
signal timing oos suppress-all 1
signal keepalive 65535
!
!Define virtual interface for multicast fast switching
!Routers joining the same session should have the virtual interfaces
!on different subnets. Otherwise packets will not be switched to the IP network
!
interface vif1
ip address 1.1.1.1 255.255.255.0
ip pim dense-mode
!
!Configure voice ports. 
!Use voice class tag 1 for port that is allowed to speak
!Use voice class tag 2 for listen-only port
!Set music threshold to make VAD less sensitive. Only noise above
!-30 dB is considered voice.
!Tie voice port to phone number 111, joining multicast session 237.111.0.0:22222.
!Joining session 111
!
voice-port 2/0/0
voice-class permanent 1
vad
connection trunk 111
music-threshold -30
operation 4-wire
!
!Joining session 111 in receive-only mode
!
voice-port 2/0/1
voice-class permanent 2
vad
connection trunk 111
music-threshold -30
operation 4-wire
!

The connection-trunk connection type is a point-to-point connection, similar to a tie-line on a PBX network. All voice traffic—including signaling—placed at one end is immediately transferred to the other.

The voice port must be configured for 4-wire operation.

High-Density Voice Modules

A multiflex trunk interface card in a high-density voice network module requires special voice-port configuration. First, select the card to configure:

voice-card 6
 codec complexity high
!

Note Codec complexity must be high. Voice multicasting does not support medium complexity, which is the default.

The following commands define the T1 channel and signaling method, and map each DS0 to voice port slot/port:ds0-group:

controller T1 6/0
 ds0-group 1 timeslots 1 type e&m-immediate-start
 ds0-group 2 timeslots 2 type e&m-immediate-start
 ds0-group 3 timeslots 3 type e&m-immediate-start
 
 ds0-group 22 timeslots 22 type e&m-immediate-start
 ds0-group 23 timeslots 23 type e&m-immediate-start

The following commands configure the voice ports on the multiflex trunk interface card:

!
voice-port 6/0:1
 connection trunk 999
!
voice-port 6/0:2
 connection trunk 999
!
voice-port 6/0:3
 connection trunk 999
 
voice-port 6/0:22
 connection trunk 999
!
voice-port 6/0:23
 connection trunk 999

Dial Peer Configuration

Cisco IOS software uses objects called dial peers to tie together telephone numbers, voice ports, and other call parameters. Configuring dial peers is similar to configuring static IP routes—you are telling the router what path to follow to route the call.

Dial peers are identified by numbers, but to avoid confusing these numbers with telephone numbers, they are usually referred to as tags. Dial peer tags are integers that can range from 1 to 2³¹ -1 (2147483647). Dial peers on the same router must have unique tags, but you can reuse the tags on other routers.

The following commands configure a dial peer with tag 1 for this voice port:

!Configure dial peer.
!Conference 1.
!Phone number 111.
!Multicast address 237.111.0.0, udp port 22222.
dial-peer voice 1 voip
destination-pattern 111
session protocol multicast
session target ipv4:237.111.0.0:22222
ip precedence 5
 codec g711ulaw
!

Tips

The destination pattern 111 for the VoIP dial peer matches the connection trunk string for the corresponding voice port.

The session protocol multicast command is essential for voice multicasting.
The session target for voice multicasting dial peers is a multicast address in the range
224.0.1.0 to 239.255.255.255. This session target must be the same for all ports in a
session. The audio RTP port is an even number in the range 16384 to 32767, and must
also be the same for all ports in a session.
The following codec restrictions:
- You must configure the same codec on all dial peers in a session.
- Only G.711, G.726, and G.729 codecs are supported.
- When the default codec, G.729, is used, it does not appear in the configuration.
Voice activity detection (VAD) is enabled by default. This setting should not be changed.

Ethernet Configuration

Configure the router's Ethernet interface as follows:

!Configure physical interface for transmitting multicast packets.
!
interface ethernet 0/0
ip address 1.5.13.13 255.255.255.0
ip pim sparse-dense-mode
ip sap listen
no shutdown
!

Note PIM should always be configured for sparse-dense-mode.

Configuring the Second Router

In router "Costello", the E&M phone is connected to voice port 3/1/1. Router "Costello" uses the same configuration as "Abbott", except for the following differences:

The virtual interface must be on a different subnet from the first router.
The IP address in the Ethernet configuration must be different from that of the first router.
The voice port and slot should match the router's hardware configuration.

Figure 2: Voice Multicasting over a LAN (Second Router)

Note The multicast session for this port, shown in the session target command, must match the multicast session configured on the first router.

The codec configured for this dial peer must match the codec for the dial peer on the first router.

Both routers must be configured to use the same connection trunk and destination pattern.

Checking the Configuration

If you have configured your routers by following these examples, you should now be able to talk over the telephones. You can also use the show dial-peer voice command on each router to verify that the data you configured is correct.

To verify that an audio path has been established, use the show call active voice command. This command displays all active voice calls traveling through the router.

Voice Multicasting over a WAN Example

The configuration for voice multicasting sessions over IP on a Frame Relay, ATM, or other WAN is the same as for the Ethernet LAN in the last example. Configure the WAN interface on each router with the ip address and ip pim sparse-dense-mode commands as shown in Voice Multicasting over an Ethernet LAN Example.

Quality of Service

Voice traffic is much more sensitive to timing variations than data traffic. For good voice performance, configure your data network so that voice packets are not lost or delayed. The following example shows one way to improve quality of service (QoS) for voice multicasting over a Frame Relay connection:

!Configure physical interface for transmitting multicast packets.
!Listen to packets of Session Announcement Protocol.
!This example uses a subinterface
!
interface serial0/0
 encapsulation frame-relay
 frame-relay traffic-shaping
 no frame-relay broadcast-queue
!
interface serial0/0.1 point-to-point
 ip address 5.5.5.5 255.255.255.0
 ip pim sparse-dense-mode
 frame-relay class hootie
 frame-relay interface-dlci 100
 frame-relay ip rtp header-compression
!
!Frame relay class commands.
!
map-class frame-relay hootie
 frame-relay cir 64000
 frame-relay bc 2000
 frame-relay mincir 64000
 no frame-relay adaptive-shaping
 frame-relay fair-queue
 frame-relay fragment 80
 frame-relay ip rtp priority 16384 16383 64

Note In the frame-relay ip rtp priority command, the first number is the audio port. The second number is the number of consecutive audio ports to which the IP RTP priority queuing applies. The third number is the bandwidth, which should equal the bandwidth needed for each call multiplied by the number of calls.

Command Reference

This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.1 command reference publications.

session protocol multicast

session protocol multicast

To set the session protocol as multicast, use the session protocol multicast dial-peer configuration command. To negate this command and return to the cisco default session protocol, use the no version of this command.

session protocol multicast

no session protocol multicast

Syntax Description

There are no keywords or arguments.

Defaults

When this command is not implemented, the default session protocol is cisco.

Command Modes

Dial-peer configuration

Command History

Release Modification

12.1(2)XH

This command was introduced.

Release	Modification
12.1(2)XH	This command was introduced.

Usage Guidelines

Use the session protocol multicast dial-peer configuration command for voice conferencing in a Hoot and Holler networking implementation. This command allows more than two ports to join the same session simultaneously. It is supported on Cisco 2600 and Cisco 3600 series routers.

Examples

The following example shows the use of the the session protocol multicast dial-peer configuration command:

Router(config)# dial-peer voice tag voip
Router(config-dial-peer)# destination-pattern phone-number
Router(config-dial-peer)# session protocol multicast
Router(config-dial-peer)# session target ipv4:address:port
Router(config-dial-peer)# ip precedence number
Router(config-dial-peer)# codec {g711alaw | g711ulaw |
g726r32 | g729ar8 | g729r8}

Related Commands

Command Description

session target ipv4

Assigns the session target for voice multicasting dial peers.

Command	Description
session target ipv4	Assigns the session target for voice multicasting dial peers.

Glossary

CIR—Committed information rate. The average rate of information transfer a subscriber (for example, the network administrator) has stipulated for a frame relay PVC.

CODEC—Coder-decoder. Device that typically uses pulse code modulation (PCM) to transform analog signals into a digital bit stream and digital signals back into analog signals. In Voice over IP, it specifies the voice coder rate of speech for a dial peer.

Dial-peer—an addressable call endpoint. In Voice over IP, there are 2 kinds of dial peers:

POTS—connected through a traditional telephony network and points to a particular voice port on a voice network device.
VoIP—connected through a packet network (IP network for VoIP) and points to specific VoIP device.

E&M—Ear and Mouth. A type of signalling traditionally used in the telecommunications industry. Indicates the use of a handset that corresponds to the ear (receiving) and mouth (transmitting) component of a telephone.

FXO—Foreign Exchange Office. An FXO interface connects to the Public Switched Telephone Network's (PSTN) central office and is the interface offered on a standard telephone. Cisco's FXO interface is an RJ-11 connector that allows an analog connection at the PSTN's central office or to a station interface on a PBX.

FXS—Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. Cisco's FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.

Hoot and Holler—A broadcast audio network used extensively by the brokerage industry for market updates and trading. Similar networks are used in publishing, transportation, power plants, and manufacturing.

IVR—Interactive Voice Response.

PBX—Private Branch Exchange. Digital or analog telephone switchboard or switching facility located on the subscriber premises and used to connect private and public telephone networks.

PVC—Permanent Virtual Circuit.

QoS—Quality of Service. QoS refers to the measure of service quality provided to the user.

Trunk—Service that allows quasi-transparent connections between two PBXs, a PBX and a local extension, or some other combination of telephony interfaces to be permanently conferenced together by the session application and signaling passed transparently through the IP network.

VAD—Voice Activation Detection.

VIC—Voice Interface Card.

VNM—Voice Network Module.

VoIP—Voice over Internet Protocol.

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