Cisco Systems Computer Accessories 3600 User Manual

Cisco Hoot and Holler over IP  
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.  
Four-wire E&M, E1/T1, FXO, and FXS configurations provide continuous VoIP connections across a  
packet network using the connection-trunk mechanism. By using the inherent point-to-multipoint  
connectivity of IP multicast (IPmc), the routers can take several inbound voice streams from the  
traditional hoot devices and forward the packetized voice over the IP network to all parties within a  
defined hoot and holler group.  
This feature module describes the Cisco Hoot and Holler over IP feature and contains the following  
sections:  
Feature Overview  
Hoot and holler networks provide "always on" multiuser conferences without requiring that users dial  
into a conference. These 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. These networks functioned as  
crude, do-it-yourself, business-to-business intercom systems.  
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 morning reports.  
Users of various forms of hoot and holler networks now include brokerages, news agencies, publishers,  
weather bureaus, transportation providers, power plant operators, manufacturers, collectibles dealers,  
talent agencies, and nationwide salvage yard organizations.  
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Feature Overview  
Hoot and holler networks are typically spread over four to eight sites, although financial retail networks  
may have hundreds of sites interconnected. Within a site, bridging (mixing voice signals) is done locally  
with a standard analog or digital bridge that may be part of a trading turret system. Between sites, there  
are two prevalent methods for providing transport:  
Point-to-point leased lines with customer-provided audio bridging at a central site, and  
Carrier-provided audio bridging.  
When customers provide their own bridging services with point-to-point leased lines, branch offices in  
a metropolitan area commonly have 25 to 50 lines or more.  
The second method, carrier-provided audio bridging, is prevalent within the United States but rare for  
overseas transport. In this scenario, the audio bridges are located at the carrier's central office and the  
4-wire lines are terminated at the client's site on a local audio-bridge equipped with 4-wire plug-ins,  
which then feed to local PA system speakers. Customer-provided hoot bridging services can now be  
replaced with Cisco Hoot and Holler over IP solutions.  
An Overview of Cisco Hoot and Holler over IP  
Cisco's VoIP technology, which was initially focused on traditional PBX toll-bypass applications, can  
be used to combine hoot and holler networks with data networks. While some customers may have done  
some level of hoot and data integration in the late 1980s with time-division multiplexing (TDM), this  
form of integration does not allow for dynamic sharing of bandwidth that is characteristic of VoIP. This  
dynamic sharing of bandwidth is even more compelling with hoot and holler than with a toll-bypass  
application, because some hoot circuits may be active for an hour or two for morning reports but might  
be dead for the rest of the day—the idle bandwidth can be used by the data applications during these  
long periods of inactivity.  
Beginning with the Cisco IOS Release 12.1(2)XH, Cisco Hoot and Holler over IP can be implemented  
using Cisco's VoIP technology. This solution leverages Cisco's IOS expertise in VoIP, quality of service  
(QoS), and IP multicasting (IPmc) and is initially available on Cisco 2600 and 3600 series multiservice  
routers.  
Figure 2 shows a diagram of the Cisco Hoot and Holler over IP solution connecting legacy hoot  
equipment over an IP network.  
Note  
The "V" on the Cisco router icons signifies that some of the hoot and holler bridging  
function is being done by the router's digital signal processors (DSPs).  
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Feature Overview  
Figure 2  
Hoot and Holler over IP using Cisco 2600 and Cisco 3600 Series Routers  
PBX  
V
FXO  
Multicast group 1  
E&M phones  
V
PBX  
PBX  
FXO  
Multicast group 3  
Multicast group 2  
V
V
FXS  
FXO  
V
T1/E1  
V
T1/E1  
Turret  
Turret  
Turret  
E&M = ear and mouth  
Four-wire E&M, E1/T1, FXO, and FXS configurations provide continuous VoIP connections across a  
packet network. By using the inherent point-to-multipoint characteristic of IPmc, the routers can take  
several inbound voice streams from the traditional hoot devices, and forward the packetized voice over  
the IP network to all parties within a defined hoot and holler group.  
Voice Multicasting  
The voice multicasting feature on Cisco 2600 and Cisco 3600 series routers uses Cisco Voice over IP  
(VoIP) technology to create a point-to-multipoint hoot and holler network over an IP connection.  
Voice multicasting telephones can be connected to routers in the following ways:  
Connect a 4-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 an FXS voice interface card that is installed in a voice network  
module.  
Connect a conventional telephone to a PBX that is connected through a E1/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 E1/T1 line per high-density voice  
network module.  
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Feature Overview  
IP/TV Access  
The Cisco Hoot and Holler over IP feature enables you to access ongoing IP/TV multicasts for listening  
to voice content of the IP/TV session. For complete information on IP/TV, see the IP/TV Content  
Manager Installation and User Guide .  
Figure 3  
Cisco Hoot and Holler over IP Access to IP/ TV Multicast  
IP/TV  
Content Manager  
IP/TV  
server  
IP/TV  
viewer  
For the Cisco Hoot and Holler over IP and IP/TV interaction to work correctly:  
Ensure that you have a properly connected and configured network for Voice over IP with the  
Cisco Hoot and Holler over IP feature enabled, using the session protocol multicast command.  
Ensure that the server configured with the IP/TV Content Manager is in the same Ethernet network  
as the Cisco Hoot and Holler over IP functionality.  
Ensure that the Cisco Hoot and Holler over IP multicast details are registered with the IP/TV  
Content Manager.  
Note  
IP/TV support for Cisco Hoot and Holler over IP uses only G.711 u-law (mu-law)  
encoding.  
IP/TV supports one audio stream for Cisco Hoot and Holler over IP.  
IP/TV does not support arbitration and mixing.  
Content Manager  
On the configuration screen (Administration Tool>Scheduled Programs>New  
Program>Configuration), provide the following details:  
Multicast address  
RTP port—defined by the dial-peer in the router  
IP/TV server—IP address or name  
From the Settings>Content Manager option:  
Click Add New.  
Enter the IP/TV server name.  
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Feature Overview  
The port number must be 80, because it is HTTP.  
Click OK and exit.  
Note  
In the Content Manager, be sure to specify the multicast IP address and RTP port for the  
Cisco Hoot and Holler over IP session.  
Interactive Voice Response (IVR)  
The Cisco Hoot and Holler over IP feature can support Interactive Voice Response (IVR) as a means of  
authentication, authorization, and accounting (AAA) control. See the section “Configuring Interactive  
Voice Response (IVR) ” in the Multiservice Applications Configuration Guide and the command  
descriptions in the Multiservice Applications Command Reference for more information.  
Migration Strategy  
To aid troubleshooting and allow for regionalized hoot and holler conferences, most hoot and holler  
networks today are structured by interconnecting multiple, regional hoot networks with a centralized  
bridge. The regional hoot networks are built using either carrier-based multidrop circuits or point-to-  
point circuits bridged by the customer. All of these circuits are connected through patch panels that  
allow for these regional bridges to be connected for a larger corporate-wide conference call. This is  
typically done for the "morning call" that is broadcast to all locations, advising of market movements,  
recommendations, and commentary. Later in the day, the patch panel may be reconfigured to allow for  
local or regional conference bridges. This allows for multiple conference calls for various purposes,  
without provisioning multiple circuits. By segmenting the network into regions, troubleshooting is also  
easier because any audio disturbance, feedback, or level problems can be isolated to a smaller subset of  
remote offices for more specific troubleshooting.  
The highly segmented nature of existing hoot and holler networks can be leveraged in the migration  
from legacy hoot technology to Cisco Hoot and Holler over IP. A small segment of the hoot network  
can be converted to Cisco Hoot and Holler over IP while preserving the operational procedures at the  
main office.  
Note that the migration to Cisco Hoot and Holler over IP does not require replacing end-user equipment  
or central bridging equipment. The main impetus for this first phase of migration is to eliminate the  
recurring expense of carrier multidrop circuits or dedicated leased lines. By minimizing changes  
presented to the end user while realizing an attractive payback period on the capital costs, migration  
success is maximized.  
As the entire hoot network converges with the data network, additional functionality can be introduced.  
Since the hoot and holler connections are now carried in standard multicast RTP packets, hoot channels  
can now be received by a soft client such as IP/TV, which can receive an IP multicast RTP stream. An  
alternate migration strategy is to use Cisco Hoot and Holler over IP technology initially as a backup for  
the existing hoot circuits within a region with a phased plan of cutting over to Cisco Hoot and Holler  
over IP as the primary transport while keeping the existing circuits as a backup for a predefined burn-in  
period.  
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Feature Overview  
Technical Details of the Cisco Hoot and Holler over IP Solution  
This section describes how Cisco Hoot and Holler over IP works from a technical perspective. It covers  
design considerations in terms of IOS configur  
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Feature Overview  
is handled by the onboard DSPs on each voice-card (NM-1V, NM-2V or NM-HDV). Arbitration  
involves identifying the various sources of the voice stream, and mixing involves taking some of those  
voice streams and combining them into a single-sourced voice stream. Cisco Hoot and Holler over IP  
can handle many inbound voice streams, but it only arbitrates and mixes three streams to be heard  
within the Hoot group. This value works fine in most applications, because beyond three streams two  
things happen in normal human conversation:  
1. People are not able to distinguish the content of more than three streams.  
2. People normally stop speaking if they hear others talking ahead of them.  
Note  
The mixing functionality does not do a summation of the voice streams.  
As previously mentioned, the DSPs in Cisco Hoot and Holler over IP do mixing for up to three streams.  
This fact is important when network administrators consider how much bandwidth they should plan for  
in their Cisco Hoot and Holler over IP network. This is especially crucial when planning WAN  
bandwidth, which is often much more expensive and much less available than LAN bandwidth.  
The advantage to this functionality is that a network administrator never has to be concerned about  
provisioning voice bandwidth for more than three times per call bandwidth for each WAN site, which  
helps to simplify overall network planning.  
Bandwidth Planning  
Four main factors must be considered with regard to bandwidth planning for  
Cisco Hoot and Holler over IP:  
1. Codecs used for VoIP (G.711, G.726, G.729 and G.729a are currently supported).  
2. Bandwidth management techniques.  
3. The number of voice streams to be mixed.  
4. The amount of guaranteed bandwidth available on the IP network. This includes both LAN and  
WAN bandwidth, and should take into consideration things such as Frame Relay CIR.  
Codecs  
By default, Cisco IOS sends all VoIP traffic (media, using RTP) at a rate of 50 packets per second. The  
packets include not only the voice sample, but also an IP, UDP, and RTP header. The IP/UDP/RTP  
header adds an additional 40 bytes to each packet. The amount of bandwidth each VoIP call consumes  
depends on the codec selected. The resulting bandwidths can be:  
G.729 or G.729a = 3000bytes * 8 bits = 24Kb/call  
G.726 = 6000bytes * 8 bits = 48Kb/call  
G.711 = 10000bytes * 8 bits = 80Kb/call  
In addition to these calculations, network administrators must consider the Layer 2 headers  
(Frame Relay, PPP, Ethernet, and so on.) and add the appropriate number of bytes to each packet.  
The following table, Table 1, assumes a payload size (bytes) of 20 ms samples per packet with  
50 packets per second.  
The value of n is equal to the number of voice streams in a session.  
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Feature Overview  
The uncompressed bandwidth includes IP/UDP/RTP headers (40 bytes) in the bandwidth calculation.  
Compressed RTP (cRTP) reduces the IP/UDP/RTP headers to between 2 to 4 bytes per packet. The  
calculation of compressed bandwidth below uses 4 bytes for a compressed IP/UDP/RTP header per  
packet.  
Maximum RTCP bandwidth is five percent of the total RTP traffic in a hoot and holler session. Since  
the Cisco Hoot and Holler over IP application supports mixing of a maximum of three voice streams,  
the RTCP bandwidth is limited to five percent of three-voice-stream traffic.  
In addition to the above, Layer 2 headers (Frame Relay, PPP, Ethernet, and so on) should be considered  
and added to the bandwidth calculation.  
Table 1  
Bandw idth Consideration Table  
RTCP Bandwidth per  
Payload Size  
(byte)  
Bandwidth/ Voice Stream  
(Kbps)  
Cisco Hoot and Holler over IP Example—One Voice Stream in a  
Codec  
Session (Kbps)  
Session (Bandwidth in Kbps)  
Uncompressed  
Compressed  
=(1)*n+(3)  
27.6  
=(2)*n+(3)  
13.2  
g.729  
g.726  
g.711  
20  
80  
24  
48  
80  
9.6  
3.6  
33.6  
65.6  
7.2  
55.2  
40.8  
160  
12.0  
92.0  
77.6  
cRTP, Variable-Payload Sizes and VAD  
Some network administrators may consider this amount of bandwidth per call unacceptable or outside  
the limits for which they can provide bandwidth, especially in the WAN. There are several options that  
network administrators have for modifying the bandwidth consumed per call:  
1. RTP header compression (cRTP)  
2. Adjustable byte-size of the voice payload  
3. Voice activity detection (VAD)  
IP/UDP/RTP headers add an additional 40 bytes to each packet, but each packet header is basically  
unchanged throughout the call. cRTP can be enabled for the VoIP calls, which reduces the IP/UDP/RTP  
headers to between 2 to 4 bytes per packet.  
More detailed documentation on cRTP can be found on CCO at:  
In addition to reducing the IP/UDP/RTP headers per packet, the network administrator also has the  
option of controlling how much voice payload is included in each packet. This is done using the bytes  
keyword and argument in a VoIP dial-peer. The following example shows a dial-peer configuration:  
dial-peer voice 1 voip  
destination-pattern 4085551234  
codec g729r8 bytes 40  
session protocol multicast  
session target ipv4:239.10.108.252:20102  
As the number of bytes per packet is modified, so too is the number of packets per second that are sent.  
Voice activity detection (VAD) enables the DSPs to dynamically sense when there are pauses in a  
conversation. When these pauses occur, no VoIP packets are sent into the network. This significantly  
reduces the amount of bandwidth used per VoIP call, sometimes as much as 40 to 50 percent. When  
voice is present, then VoIP packets are again sent. When using Cisco Hoot and Holler over IP, VAD must  
be enabled to reduce the amount of processing of idle packets by the DSPs. In basic VoIP, VAD can be  
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Feature Overview  
enabled or disabled, but since the DSPs also have to do arbitration and mixing, VAD must be disabled  
to reduce the DSPs processing load. In addition to enabling VAD (which is on by default), network  
administrators should modify the VAD parameters that sense background noise so that idle noise does  
not consume bandwidth.  
This can be configured as in the following E&M port example:  
voice class permanent 1  
signal timing oos timeout disabled  
signal keepalive 65535  
!
voice-port 1/0/0  
voice-class permanent 1  
connection trunk 111  
music-threshold -30  
operation 4-wire  
The configuration above is used for a voice-port that is in send/receive mode, and only noise above  
-30Db is considered voice.  
Virtual Interface (Vif)  
In all Cisco Hoot and Holler over IP implementations, the routers are configured with an "interface  
vif1." This is a virtual interface that is similar to a loopback interface—a logical IP interface that is  
always up when the router is active. In addition, it must be configured so 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 must reside on its own unique subnet, and that subnet should be included in the routing  
protocol updates (RIP, OSPF, and so on).  
Connection Trunk  
Cisco Hoot and Holler over IP provides an "always-on" communications bridge—end users do not need  
to dial any phone numbers to reach the other members of a hoot group. To simulate this functionality,  
Cisco IOS provides a feature called "connection trunk." Connection trunk provides a permanent voice  
call, without requiring any input from the end user, because all of the digits are internally dialed by the  
router/gateway.  
With traditional VoIP usages of connection trunk, the call is mapped to a remote router/gateway, and all  
the H.323 signaling is handled dynamically when the trunk is established. With  
Cisco Hoot and Holler over IP, the connection trunk is established to the IP address of the IP multicast  
(IPmc) group that maps to the hoot group.  
In addition, all negotiation of UDP ports for the audio stream is manually configured. The following  
example shows an E&M voice port connection trunk set up for Cisco Hoot and Holler over IP:  
voice-port 1/0/0  
connection trunk 111  
music-threshold -30  
operation 4-wire  
!
dial-peer voice 1 voip  
destination-pattern 111  
voice-class permanent 1  
session protocol multicast  
session target ipv4:237.111.0.0:22222  
ip precedence 5  
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Feature Overview  
In this example, the digits in the connection trunk 111 string match the destination pattern of the VoIP  
dial peer. Also, the session protocol is set to multicast and the session target is pointing to the IPmc  
group number with the UDP port (22222) predefined.  
Benefits  
Cisco Hoot and Holler over IP provides the following benefits:  
Eliminates yearly reoccurring circuit-switched telecom charges (toll-bypass).  
Eliminates the need for leased-lines and the accompanying charges.  
Reduces the need for Hoot and Holler bridges.  
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 turret support).  
Ability to integrate voice, video, and data signaling capabilities.  
Related Documents  
For information about installing voice network modules and voice interface cards in Cisco 2600 and  
Cisco 3600 series routers, see the following publications:  
Cisco Network Module Hardware Installation Guide  
WAN Interface Card Hardware Installation Guide  
For information about configuring Voice over IP features, see the following publications:  
Software Configuration Guide for Cisco 3600 Series and Cisco 2600 Series Routers  
Voice over IP Quick Start Guide  
Cisco IOS Multiservice Applications Configuration Guide , Release 12.1  
For further information about IP multicasting, go to this site:  
IP Multicast Site (http://www.cisco.com/ipmulticast)  
For further information about IP/TV, see the following publication:  
IP/TV Content Manager User Guide  
For further information about interactive voice response (IVR), see the following document:  
Configuring Interactive Voice Response for Cisco Access Platforms  
Restrictions  
Cisco Hoot and Holler over IP supports the mixing of only three voice streams.  
IP/TV does not support the mixing of audio streams.  
IP/TV supports only G.711 u-law (mu-law).  
Voice Interface Card Basic Rate Interface (VIC-BRI) is not supported.  
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Supported Platforms  
Supported Platforms  
Router Platforms:  
Cisco 2600  
Cisco 3600 series  
Network Modules:  
NM-HDV  
NM-1V  
NM-2V  
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.  
To obtain lists of MIBs supported by platform and Cisco IOS release and to download MIB modules,  
go to the Cisco MIB web site on Cisco Connection Online (CCO) at  
RFCs  
No new or modified RFCs are supported by this feature.  
Configuration Tasks  
See the following sections for configuring Cisco Hoot and Holler over IP:  
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Configuration Tasks  
Configuring Multicast Routing (Required)  
To enable multicast routing on the platform, perform the following steps:  
Command  
Purpose  
Router(config)# ip multicast-routing  
Step 1  
Enables multicast routing.  
Configuring the Virtual Interface (Required)  
To configure the virtual interface for multicast fast switching, perform the following steps:  
Command  
Purpose  
Router(config)# interface vif1  
Step 1  
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.  
Router(config-if)# ip address address subnet-mask  
Step 2  
Step 3  
Assigns the IP address and subnet mask for the  
virtual interface.  
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 VoIP Dial Peers (Required)  
To configure the VoIP dial peers on the router, perform the following steps:  
Command  
Purpose  
Router(config)# dial-peer voice tag voip  
Step 1  
Step 2  
Assigns a variable number (tag) to the VoIP dial  
peer.  
Router(config-dial-peer)# destination-pattern  
multicast-session-number  
The destination pattern for the VoIP dial peer must  
match the value of the multicast-session-number  
string for the corresponding voice port.  
Router(config-dial-peer)# session protocol multicast  
Step 3  
Step 4  
This step is mandatory for voice multicasting and  
is the command introduced specifically for the  
Cisco Hoot and Holler over IP application.  
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. An odd-numbered port  
(UDP port number + 1) is used for the RTCP  
traffic for that session.  
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Command  
Purpose  
Router(config-dial-peer)# ip precedence number  
Step 5  
Step 6  
(Optional) Specifies the IP precedence.  
Router(config-dial-peer)# codec (codec-type}  
Configures the codec. You must configure the  
same codec on all dial peers in a session.  
When the default codec, g729r8, is used, it does  
not appear in the configuration when the show  
running command is used.  
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Configuration Tasks  
Configuring E&M Voice Ports (Required, if used)  
If using E&M voice ports, configure them by performing the following steps:  
Command  
Purpose  
Router(config)# voice class permanent tag1  
Step 1  
Step 2  
Defines voice class for transmit-receive mode.  
Disables signaling loss detection.  
Router(config-class)# signal timing oos timeout  
disabled  
Router(config-class)# signal keepalive number  
Step 3  
Step 4  
Step 5  
Specifies keepalive signaling packet interval.  
Returns to global configuration mode.  
Selects the voice port to configure.  
Router(config-class)# exit  
Router(config)# voice-port  
router-slot/voice-slot/VIC-port  
Router(config-voice-port)# voice-class permanent tag1  
Step 6  
Uses voice class tag1 for the port that is allowed to  
speak.  
Router(config-voice-port)# connection trunk  
multicast-session-number  
Step 7  
Step 8  
Ties the voice port to a multicast-session number.  
Router(config-voice-port)# music-threshold threshold  
(Optional) Sets the music threshold to make VAD  
less sensitive.  
Router(config-voice-port)# operation 4-wire  
Step 9  
Specifies 4-wire operation. (2-wire is the default.)  
Router(config-voice-port)# type { 1 | 2 | 3 | 5 }  
Step 10  
Selects the appropriate E&M interface type  
(depending on the end connection—such as PBX):  
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  
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Configuration Tasks  
Command  
Purpose  
Router(config-voice-port)# signal { wink-start |  
immediate | delay-dial }  
Step 11  
Configures the signaling type for E&M voice  
ports. The default is wink-start. Select  
immediate for the Cisco Hoot and Holler over IP  
application. In the immediate-start protocol, the  
originating side does not wait for a wink before  
sending addressing information. After receiving  
addressing digits, the terminating side then goes  
off-hook for the duration of the call. The  
originating endpoint maintains off-hook for the  
duration of the call.  
Router(config-voice-port)# voice-port  
router-slot/voice-slot/VIC-port  
Step 12  
Selects another voice port.  
Router(config-voice-port)# voice-class permanent tag2  
Step 13  
Step 14  
Uses voice class tag2 for the receive-only port.  
Router(config-voice-port)# connection trunk  
multicast-session-number  
Ties the voice port to the same multicast-session  
number as in Step 12.  
Router(config-voice-port)# music-threshold threshold  
Step 15  
Step 16  
(Optional) Sets the music threshold to make VAD  
less sensitive.  
Router(config-voice-port)# operation 4-wire  
Specifies 4-wire operation. (2-wire is the default.)  
Configuring for Receive Only Mode (Optional)  
To configure Cisco Hoot and Holler over IP as receive-only mode, perform the following steps:  
Command  
Purpose  
Router(config-class)# voice class permanent tag2  
Step 1  
Step 2  
Step 3  
Defines voice class for receive-only mode.  
Specifies the received signal pattern.  
Router(config-class)# signal pattern oos receive 0000  
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.  
Router(config-class)# signal keepalive number  
Step 4  
Specifies keepalive signaling packet interval.  
Configuring Relevant Interface (Serial/Ethernet) (Required)  
To configure either the serial or Ethernet interface, perform the following steps:  
Command  
Purpose  
Router(config)# interface { ethernet | serial }  
slot/port  
Step 1  
Step 2  
Configures the physical interface (serial/Ethernet)  
for transmitting multicast packets.  
Router(config-if)# ip address address subnet-mask  
Assigns the IP address and subnet mask for the  
interface.  
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Configuration Examples  
Configuring Voice Ports in High-Density Voice Network Modules (Required, if  
using T1/E1)  
A multiflex trunk interface card (NM-HDV) in a high-density voice network module requires special  
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Note  
In all of the following configuration examples, the routers are configured with an  
interface vif1 command. 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, and so on).  
Voice Multicasting over an Ethernet LAN  
Figure 4 shows the simplest configuration for voice multicasting over an Ethernet LAN. Two routers  
are connected to each other over the Ethernet LAN. One E&M phone is connected to each router.  
Figure 4  
Voice Multicasting over a LAN  
E&M VIC  
3/1/1  
IP cloud  
E&M VIC  
2/0/0  
Router 2  
Router 1  
In router Abbott (Figure 4), the phone is connected to voice port 2/0/0, using the  
router-slot/voice-slot/VIC-port numbering convention. This router is configured as in the following  
example:  
hostname Abbott  
!
ip multicast-routing  
!
voice class permanent 1  
signal timing oos timeout disabled  
signal keepalive 65535  
!
interface Vif1  
ip address 1.1.1.1 255.0.0.0  
ip pim sparse-dense-mode  
!
interface Ethernet0/0  
ip address 3.3.3.1 255.0.0.0  
ip pim sparse-dense-mode  
!
ip route 2.0.0.0 255.0.0.0 3.3.3.2  
!
voice-port 2/0/0  
voice-class permanent 1  
connection trunk 111  
operation 4-wire  
!
dial-peer voice 1 voip  
destination-pattern 111  
session protocol multicast  
session target ipv4:237.111.0.111:22222  
!
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Note  
Note  
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 E&M voice port must be configured for 4-wire operation.  
Configuring the Second Router  
In router Costello (Figure 4), 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.  
hostname Costello  
!
ip multicast-routing  
!
voice class permanent 1  
signal timing oos timeout disabled  
signal keepalive 65535  
!
!
interface Vif1  
ip address 2.2.2.2 255.0.0.0  
ip pim sparse-dense-mode  
!
interface Ethernet0/0  
ip address 3.3.3.2 255.0.0.0  
ip pim sparse-dense-mode  
!
ip route 1.0.0.0 255.0.0.0 3.3.3.1  
!
voice-port 3/1/1  
voice-class permanent 1  
timeouts wait-release 3  
connection trunk 222  
music-threshold -30  
operation 4-wire  
!
dial-peer voice 1 voip  
destination-pattern 111  
session protocol multicast  
session target ipv4:237.111.0.111:22222  
!
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.  
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Configuration Examples  
Verifying 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.  
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 show how to 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 show how to configure the voice ports on the multiflex trunk interface card:  
!
voice-port 6/0:1  
connection trunk 111  
!
voice-port 6/0:2  
connection trunk 111  
!
voice-port 6/0:3  
connection trunk 111  
voice-port 6/0:22  
connection trunk 111  
!
voice-port 6/0:23  
connection trunk 111  
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  
instructing the router what path to follow to route the call.  
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Configuration Examples  
Dial peers are identified by numbers, but to avoid confusing these numbers with telephone numbers,  
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Configuration Examples  
Voice Multicasting over a WAN  
The configuration for voice multicasting sessions over IP on Frame Relay is the same as for the Ethernet  
LAN in the previous example. Configure the WAN interface on each router with the ip address and ip  
pim sparse-dense-mode commands as shown in the section, Voice Multicasting over an Ethernet LAN.  
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 (SAP).  
!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.  
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Cisco Hoot and Holler over IP with Ethernet Topology (Two Hoot Groups)  
Figure 5  
Cisco Hoot and Holler over IP w ith Ethernet Topology  
E&M  
4-wire  
FXS  
Router 2  
Router 1  
Router 3  
FXO  
Group 1  
Group 2  
FXO  
PBX  
PBX  
E&M = ear and mouth  
In this configuration, two hoot and holler groups are set up by defining two multicast groups  
(237.111.0.111 and 237.111.0.112) and mapping the connection trunk 111 and connection trunk 112  
commands from the voice ports to the VoIP dial peers associated with each group. Each router is  
connected to a dedicated switch port, and IP precedence is set to 5 for all Cisco Hoot and Holler over  
IP packets.  
Router-1 (E&M 4-Wire Ports)  
hostname Router-1  
!
ip multicast-routing  
!
voice class permanent 1  
signal timing oos timeout disabled  
signal keepalive 65535  
!
interface Vif1  
ip address 1.1.1.1 255.255.255.0  
ip pim sparse-dense-mode  
!
!
interface Ethernet0/0  
ip address 1.5.13.1 255.255.255.0  
ip pim sparse-dense-mode  
!
router rip  
network 1.1.1.0  
network 1.5.13.0  
!
voice-port 1/0/0  
voice-class permanent 1  
connection trunk 111  
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music-threshold -30  
operation 4-wire  
!
voice-port 1/0/1  
voice-class permanent 1  
connection trunk 112  
music-threshold -30  
operation 4-wire  
!
dial-peer voice 111 voip  
destination-pattern 111  
session protocol multicast  
session target ipv4:237.111.0.111:22222  
ip precedence 5  
!
dial-peer voice 112 voip  
destination-pattern 112  
session protocol multicast  
session target ipv4:239.194.0.10:22224  
ip precedence 5  
!
end  
Router-2 (FXS Ports)  
hostname Router-2  
!
ip multicast-routing  
!
voice class permanent 1  
signal timing oos timeout disabled  
signal keepalive 65535  
!
interface Vif1  
ip address 2.2.2.2 255.255.255.0  
ip pim sparse-dense-mode  
!
interface Ethernet0/0  
ip address 1.5.13.2 255.255.255.0  
ip pim sparse-dense-mode  
!
router rip  
network 2.2.2.0  
network 1.5.13.0  
!
dial-peer voice 111 voip  
destination-pattern 111  
session protocol multicast  
session target ipv4:237.111.0.111:22222  
ip precedence 5  
!
dial-peer voice 112 voip  
destination-pattern 112  
session protocol multicast  
session target ipv4:239.194.0.10:22224  
ip precedence 5  
!
end  
Note  
If you want to join the hoot and holler session directly without having to dial any session  
numbers, use the command connection plar, followed by the multicast-session number.  
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Router-3 (FXO Ports)  
hostname Router-3  
!
ip multicast-routing  
!
voice class permanent 1  
signal timing oos timeout disabled  
signal keepalive 65535  
!
interface Vif1  
ip address 3.3.3.3 255.255.255.0  
ip pim sparse-dense-mode  
!
interface Ethernet0/0  
ip address 1.5.13.3 255.255.255.0  
ip pim sparse-dense-mode  
!
router rip  
network 3.3.3.0  
network 1.5.13.0  
!
dial-peer voice 111 voip  
destination-pattern 111  
session protocol multicast  
session target ipv4:237.111.0.111:22222  
ip precedence 5  
!
dial-peer voice 112 voip  
destination-pattern 112  
session protocol multicast  
session target ipv4:239.194.0.10:22224  
ip precedence 5  
!
end  
Cisco Hoot and Holler over IP with Frame-Relay Topology (One Hoot Group)  
Figure 6  
Cisco Hoot and Holler over IP w ith Fram e-Relay Topology  
Router 1  
IP multicast  
network  
Router 2  
Router 3  
In this topology, three routers are connected using 64K Frame-Relay PVCs in a hub and spoke topology,  
with Router 1 being the hub. We have defined one hoot and holler group. All three routers have been  
configured to traffic-shape their data and voice on the WAN to CIR, and all three routers are using IP  
RTP priority to guarantee QoS for the Cisco Hoot and Holler over IP packets. In addition, the  
frame-relay broadcast-queue is disabled on the serial interfaces. This occurs because, by default, the  
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Configuration Examples  
broadcast-queue is only 40 packets deep and Cisco Hoot and Holler over IP transmits packets at 50  
packets per second. Unless the queue is disabled, some packets would be dropped and voice QoS would  
be degraded.  
Router-1  
hostname Router-1  
!
ip multicast-routing  
!
voice class permanent 1  
signal timing oos timeout disabled  
signal keepalive 65535  
!
interface Vif1  
ip address 1.1.1.1 255.255.255.0  
ip pim sparse-dense-mode  
!
router rip  
network 1.1.1.0  
network 5.5.5.0  
network 5.5.6.0  
!
interface Serial0/0  
no ip address  
ip pim sparse-dense-mode  
encapsulation frame-relay  
frame-relay traffic-shaping  
no frame-relay broadcast-queue  
!
interface Serial0/0.1 point-to-point  
ip address 5.5.5.1 255.255.255.0  
ip pim sparse-dense-mode  
frame-relay class hoot-n-holler  
frame-relay interface-dlci 100  
frame-relay ip rtp header-compression  
!
interface Serial0/0.2 point-to-point  
ip address 5.5.6.1 255.255.255.0  
ip pim sparse-dense-mode  
frame-relay class hoot-n-holler  
frame-relay interface-dlci 101  
frame-relay ip rtp header-compression  
!
map-class frame-relay hoot-n-holler  
frame-relay cir 128000  
frame-relay bc 1280  
frame-relay mincir 128000  
frame-relay fragment 160  
frame-relay ip rtp priority 16384 16384 128  
no frame-relay adaptive-shaping  
!
voice-port 1/0/0  
voice-class permanent 1  
connection trunk 111  
music-threshold -30  
operation 4-wire  
!
dial-peer voice 1 voip  
destination-pattern 111  
session protocol multicast  
session target ipv4:237.111.0.0:22222  
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ip precedence 5  
Router-2  
hostname Router-2  
!
ip multicast-routing  
!
voice class permanent 1  
signal timing oos timeout disabled  
signal keepalive 65535  
!
interface Vif1  
ip address 2.2.2.2 255.255.255.0  
ip pim sparse-dense-mode  
!
router rip  
network 2.2.2.0  
network 5.5.5.0  
!
interface Serial0/0  
no ip address  
ip pim sparse-dense-mode  
encapsulation frame-relay  
frame-relay traffic-shaping  
no frame-relay broadcast-queue  
!
interface Serial0/0.1 point-to-point  
ip address 5.5.5.2 255.255.255.0  
ip pim sparse-dense-mode  
frame-relay class hoot-n-holler  
frame-relay interface-dlci 100  
frame-relay ip rtp header-compression  
!
map-class frame-relay hoot-n-holler  
frame-relay cir 128000  
frame-relay bc 1280  
frame-relay mincir 128000  
frame-relay fragment 160  
frame-relay ip rtp priority 16384 16383 128  
no frame-relay adaptive-shaping  
!
voice-port 1/0/0  
voice-class permanent 1  
connection trunk 111  
music-threshold -30  
operation 4-wire  
!
dial-peer voice 1 voip  
destination-pattern 111  
session protocol multicast  
session target ipv4:237.111.0.0:22222  
ip precedence 5  
Router-3  
hostname Router-3  
!
ip multicast-routing  
!
voice class permanent 1  
signal timing oos timeout disabled  
signal keepalive 65535  
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!
interface Vif1  
ip address 3.3.3.3 255.255.255.0  
ip pim sparse-dense-mode  
!
router rip  
network 3.3.3.0  
network 5.5.6.0  
!
interface Serial0/0  
no ip address  
ip pim sparse-dense-mode  
encapsulation frame-relay  
frame-relay traffic-shaping  
no frame-relay broadcast queue  
!
interface Serial0/0.1 point-to-point  
ip address 5.5.6.2 255.255.255.0  
ip pim sparse-dense-mode  
frame-relay class hoot-n-holler  
frame-relay interface-dlci 101  
frame-relay ip rtp header-compression  
!
map-class frame-relay hoot-n-holler  
frame-relay cir 128000  
frame-relay bc 1280  
frame-relay mincir 128000  
frame-relay fragment 160  
frame-relay ip rtp priority 16384 16383 128  
no frame-relay adaptive-shaping  
!
voice-port 1/0/0  
voice-class permanent 1  
connection trunk 111  
music-threshold -30  
operation 4-wire  
!
dial-peer voice 1 voip  
destination-pattern 111  
session protocol multicast  
session target ipv4:237.111.0.0:22222  
ip precedence 5  
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Command Reference  
Command Reference  
This section documents a new command. All other commands used with this feature are documented in  
the Cisco IOS Release 12.1 command reference publications.  
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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  
Defaults  
There are no keywords or arguments.  
When this command is not implemented, the default session protocol is cisco.  
Dial-peer configuration  
Command Modes  
Command History  
Release  
Modification  
12.1(2)XH  
This command was introduced on Cisco 2600 and Cisco 3600 series routers  
for the Cisco Hoot and Holler over IP application.  
Usage Guidelines  
Examples  
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.  
The following example shows the use of the the session protocol multicast dial-peer configuration  
command in context with its accompanying commands:  
Router(config)# dial-peer voice 111 voip  
Router(config-dial-peer)# destination-pattern 111  
Router(config-dial-peer)# session protocol multicast  
Router(config-dial-peer)# session target ipv4:237.111.0.111:22222  
Router(config-dial-peer)# ip precedence 5  
Router(config-dial-peer)# codec g711ulaw  
Related Commands  
Command  
Description  
session target ipv4  
Assigns the session target for voice-multicasting dial peers.  
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Glossary  
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 that contains configuration information including voice  
protocol, a codec type, and a telephone number associated with the call. There are five kinds of dial  
peers: POTS, VoIP, VoFR, VoATM, and VoHDLC. In Voice over IP, there are two 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.  
DSP—Digital signal processor.  
DTMF—Dual tone multifrequency. Uses two simultaneous voice-band tones for dial such as  
touch-tone.  
E&M—Ear and mouth. Stands for the 2-wire or 4-wire interface with separate signaling paths for the  
receiving and transmitting signals—a type of signaling 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. E&M is a trunking arrangement generally used for two-way switch-to-switch  
or switch-to-network connecting. The Cisco analog E&M interface is an RJ-48 connector that allows  
connections to PBX trunk lines (tie lines). E&M is also available on E1 and T1 digital interfaces.  
FXO—Foreign Exchange Office. An FXO interface connects to the Public Switched Telephone  
Network (PSTN) central office and is the interface offered on a standard telephone. The Cisco 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. When someone dials in, IVR responds with a prompt to get a personal  
identification number (PIN), and so on.  
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.  
TDM—Time-division multiplexing.  
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.  
VADVoice activity detection.  
VICVoice interface card.  
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Glossary  
VNMVoice network module.  
VoIPVoice over Internet Protocol.  
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