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This chapter describes Catalyst digital signaling processor (DSP) resources, with emphasis on two new Catalyst 4000 and Catalyst 6000 voice modules, and it discusses how to provision these resources. These new modules are the WS-X4604-GWY for the Catalyst 4000 and the WS-X6608-T1 (WS-X6608-E1 for countries outside the USA) for the Catalyst 6000. They are available for use with Cisco CallManager Release 3.0. They can perform conferencing and Media Termination Point (MTP) transcoding services in addition to serving as a PSTN gateway (see "Gateway Selection").
This chapter includes the following major sections:
The DSP resources on the new Catalyst 4000 and 6000 gateway modules essentially provide hardware support for IP telephony features offered by the Cisco CallManager. These features are hardware-enabled voice conferencing, hardware-based MTP support for supplementary services, and transcoding services.
Catalyst-enabled conferencing is the ability to support voice conferences in hardware. DSPs are used to convert Voice over IP (VoIP) sessions into Time-Division Multiplexing (TDM) streams, which can then be mixed into a multi-party conference call.
The Catalyst MTP service can act either like the original software MTP resource or as a transcoding MTP resource. An MTP service is the ability to provide supplementary services such as hold, transfer, and conferencing when using gateways and clients that do not support the H.323v2 feature of OpenLogicalChannel and CloseLogicalChannel with the EmptyCapabilitiesSet. MTP is available as a software feature that can run on Cisco CallManager or a separate Windows NT server. When running in software on a Cisco CallManager, 24 MTP sessions are supported. When running on a separate Windows NT server, up to 48 MTP sessions are supported. The new Catalyst gateway modules can support this same functionality, but they provide the service in hardware.
Transcoding is in effect an IP-to-IP voice gateway service. A transcoding node can convert a G.711 voice stream into a low-bit-rate (LBR) compressed voice stream, such as G.729a. This is critical for enabling applications such as integrated voice response (IVR), uOne messaging, and conference calls over slow speed IP WANs. MTP transcoding is currently supported only on the Catalyst voice gateways.
To scale IP telephony systems in large enterprise environments, hardware-based conferencing must be used. The new hardware for the Catalyst 4000 and Catalyst 6000 switch families was developed with this requirement in mind. These new Catalyst voice modules can handle conferencing in hardware, eliminating the requirement of running a software conferencing service on a Windows NT server in the IP telephony network.
The following points summarize the design capabilities and requirements of the new Catalyst voice modules:
The Catalyst 4000 module, the WS-X4604-GWY, can support up to four simultaneous conference calls of six callers each. The Catalyst 6000's T1 or E1 PSTN gateway module, the WS-X6608, also has the ability to support conferencing. After the WS-X6608 has been added as a T1 or E1 Cisco AVVID gateway, it can be configured, on a per-port basis, for conferencing services. The Catalyst 6000 conferencing module supports up to six callers per conference call with a maximum of 16 simultaneous conference callers per configured logical port. This results in a maximum of 128 conference participants per module.
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Note See Table 9-1 for a summary of conference call densities for each module. |
Both the WS-X4604-GWY and WS-X6608-T1 (or WS-X6608-E1) modules use the Skinny Station Protocol to communicate with the Cisco CallManager when providing conferencing or transcoding services. The Catalyst 6000 voice conferencing solution can support both compressed and uncompressed conference attendees. On the Catalyst 4000, only G.711, or uncompressed, calls can be joined to a conference call. When the conferencing service registers with the Cisco CallManager, using the Skinny Station Protocol, it announces that only G.711 voice calls can be connected. If any compressed calls request to be joined to a conference call, the Cisco CallManager connects them to a transcoding port first, to convert the compressed voice call to G.711. Once the G.711 connections are associated with a particular conferencing session (maximum of six participants per conference call), the call is converted to a TDM stream and passed to the summing logic, which the streams together. Unlike the WS-X6608-x1, which can mix all conference call participants, the WS-X4604-GWY will sum only the three dominant speakers.
The following additional points should also be noted:
Figure 9-1 illustrates the components used in Catalyst conferencing services.
The following caveats apply to Catalyst conferencing services:
Conference calls across an IP WAN are addressed in the next section, "Catalyst MTP Transcoding Services."
Introducing the WAN into an IP telephony implementation forces the issue of voice compression. In the previous designs shown in this document, all campus-oriented voice was uncompressed (G.711) to provide the highest quality while incurring the fewest complications. Once a WAN-enabled network is implemented, voice compression between sites is the recommended design choice. This calls into question how WAN users use the conferencing services or IP-enabled applications, such as the uOne Messaging Server, which support only G.711 voice connections. The solution is to use hardware-based MTP transcoding services to convert the compressed voice streams into G.711.
The following points summarize the design capabilities and requirements of the MTP transcoding:
Voice compression between IP phones is easily configured through the use of regions and locations in Cisco CallManager. However, both the Catalyst conferencing services and the uOne messaging software currently support only G.711, or uncompressed, connections. For these situations, MTP transcoding or packet-to-packet gateway functionality has been added to two of the new modules for the Catalyst 4000 and Catalyst 6000. A packet-to-packet gateway is a device with DSPs that has the job of transcoding between voice streams using different compression algorithms. That is, when a user on an IP phone at a remote location calls a user located at the central location, Cisco CallManager instructs the remote IP phone to use compressed voice, or G.729a, only for the WAN call. However, if the called party at the central site is unavailable, the call rolls to the uOne messaging system, which supports G.711 only. In this case, a packet-to-packet gateway transcodes the G.729a voice stream to G.711 to leave a message with the uOne Messaging Server. See Figure 9-2.
Connecting sites across an IP WAN for conference calls presents a complex scenario. In this scenario, the Catalyst modules must perform the conferencing service as well as the IP-to-IP transcoding service to uncompress the WAN IP voice connection. In Figure 9-3 a remote user joins a conference call at the central location. This three-participant conference call uses seven DSP channels on the Catalyst 4000 module and three DSP channels on the Catalyst 6000. Following is the breakdown:
Catalyst 4000:
Catalyst 6000:
H.323v2 Inter-cluster Trunks are used to connect Cisco CallManager clusters. If transcoding services are needed between clusters, then the Inter-cluster trunks are configured with MTP. In the case, all calls between clusters will be routed through the MTP/transcoding devices in each cluster. The MTP service will be used regardless of whether transcoding is needed for that particular inter-cluster call or not. Unlike previous versions, Cisco CallManager Release 3.0 supports compressed voice call connection through the MTP service provided a hardware MTP is used. Figure 9-4 shows an inter-cluster call flow.
Inter-Cluster MTP/Transcoding details:
The following summary caveats apply to Catalyst MTP transcoding:
Table 9-1 shows DSP resources that can be configured on the Catalyst voice services modules.
| Catalyst Voice Modules | PSTN Gateway Sessions | Conferencing Sessions | MTP Transcoding Sessions |
|---|---|---|---|
Catalyst 4000 WS-X4604-GWY | 104 G.711 calls | 24 G.711 conferencing calls; maximum of 4 conferences of 6 participants | 14 MTP transcoding calls |
Catalyst 6000 WS-6608-T1 or WS-6608-E1 | 32 G.711 calls per physical DS1 port; 256 per module | 16 conferencing calls per physical port, maximum conference size of 6 participants; 128 conference sessions per module | 16 MTP transcoding calls per physical port; 128 per module |
The PSTN gateway and voice services module for the Catalyst 4003 and 4006 switches, supports three analog voice interface cards (VICs) with two ports each or one T1/E1card with two ports and two analog VICs. The VIC interfaces can be provisioned in any combination of Foreign Exchange Office (FXO), Foreign Exchange Station (FXS), or Ear & Mouth (E&M). Additionally, when configured as an IP telephony gateway from the command line interface (CLI), this module can support conferencing and transcoding services.
The Catalyst 4000 voice gateway module can be configured in either toll bypass mode or gateway mode. However, the module's conferencing and transcoding resources can be configured only in gateway mode. Once the gateway mode is enabled, the module's 24 DSPs (4 SIMMs with 6 DSPs each) are automatically provisioned as follows:
Figure 9-5 shows a physical representation of the Catalyst 4000 voice gateway module in gateway mode.
Gateway mode is the default configuration. The conferencing-to-transcoding ratios can be changed from the CLI, as shown by the configuration commands. By changing the number of transcoding sessions to 14 instead of 16, an additional eight conferencing sessions can be enabled.
The following configuration notes apply to the Catalyst 4000 module:
The WS-6608-T1 (or WS-6608-E1 for European countries) is the same module that provides T1 or E1 PSTN gateway support for the Catalyst 6000, as described in "Gateway Selection." This module has eight channel associated signaling (CAS) or PRI interfaces, each of which has its own CPU and DSPs. Once the card has been added from Cisco CallManager as a voice gateway, it can be configured as a conferencing or MTP transcoding node. Each port acts independently of the other ports on the module. Specifically, each port can be configured only as a PSTN gateway interface, a conferencing node, or an MTP transcoding node. In most configurations, a transcoding node would be configured for each conferencing node.
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Note Conferencing and MTP transcoding services cannot cross port boundaries. |
Whether acting as a PSTN gateway, a conferencing resource, or an MTP transcoding resource, each port on the module requires its own IP address. The port can be configured to have either a static IP address or an IP address provided by DHCP. If a static IP is entered, a TFTP server address must also be added because the ports actually get all configuration information from the downloaded TFTP configuration file. Once configured through the Cisco CallManager interface, each port is capable of supporting one of the following configurations:
Figure 9-6 shows one possible configuration of the Catalyst 6000 voice gateway module. In this diagram, the module has two of its eight ports configured in PSTN gateway mode, three ports in conferencing mode, and three ports in MTP transcoding mode.
Posted: Thu Jul 27 10:35:27 PDT 2000
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