Cisco Unified Communications SRND Based on Cisco Unified Communications Manager 7.x
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Gateways

Table Of Contents

Gateways

What's New in This Chapter

Traffic Patterns and Gateway Sizing

Definitions and Terminology

PSTN Traffic Patterns

Normal Business Traffic Profile

Contact Center Traffic Profile

Gateway Sizing for Contact Center Traffic

Voice Activity Detection (VAD)

Codec

Performance Overload

Performance Tuning

Additional Information

TDM and VoIP Trunking Gateways

Understanding Cisco Gateways

Cisco Access Analog Gateways

Cisco Access Digital Trunk Gateways

Tuning Gateway Gain Settings

Gateway Selection

Core Feature Requirements

Gateway Protocols

Gateway Protocols and Core Feature Requirements

DTMF Relay

Supplementary Services

Unified CM Redundancy

Site-Specific Gateway Requirements

QSIG Support

Fax and Modem Support

Gateway Support for Fax Passthrough and Fax Relay

Best Practices

Super-Group 3 Fax Support

Gateway Support for Modem Passthrough and Modem Relay

Best Practices

V.90 Support

Supported Platforms and Features

Platform Protocol Support

Gateway Configuration Examples

Cisco IOS Gateway Configuration for Modem Passthrough

Cisco VG248 Configuration for Modem Passthrough

Clock Sourcing for Fax and Modem Passthrough

T.38 Fax Relay

Named Service Event (NSE) T.38 Fax Relay

Protocol-Based T.38 Fax Relay

Gateways for Video Telephony

Routing Inbound Calls from the PSTN

Routing Outbound Calls to the PSTN

Automated Alternate Routing (AAR)

Least-Cost Routing

ISDN B-Channel Binding, Rollover, and Busy Out

Inbound Calls

Outbound Calls

Configuring the Gateways in Unified CM

Call Signaling Port Numbers

Call Signaling Timers

Bearer Capabilities of Voice Gateways


Gateways


Last revised on: September 18, 2009

 

Gateways provide a number of methods for connecting an IP telephony network to the Public Switched Telephone Network (PSTN), a legacy PBX, or key systems. Gateways range from specialized, entry-level and stand-alone voice gateways to high-end, feature-rich integrated router and Cisco Catalyst gateways.

This chapter explains important factors to consider when selecting a Cisco gateway to provide the appropriate protocol and feature support for your IP Telephony network. The main topics discussed in this chapter include:

Traffic Patterns and Gateway Sizing

TDM and VoIP Trunking Gateways

Understanding Cisco Gateways

Gateway Selection

QSIG Support

Fax and Modem Support

Gateways for Video Telephony

What's New in This Chapter

Table 4-1 lists the topics that are new in this chapter or that have changed significantly from previous releases of this document.

 

Table 4-1 New or Changed Information Since the Previous Release of This Document 

New or Revised Topic
Described in:

Bearer capabilities

Bearer Capabilities of Voice Gateways

Cisco 2900 and 3900 Series Integrated Services Routers (ISRs)

Gateway Selection

Cisco VGD-1T3, VG202, and VG204 Gateways

Gateway Protocols

Site-Specific Gateway Requirements

H.320 video channel bonding

ISDN B-Channel Binding, Rollover, and Busy Out

TDM gateways

TDM and VoIP Trunking Gateways

Videoconferencing gateways

Gateways for Video Telephony

Voice gain and attenuation settings

Tuning Gateway Gain Settings


Traffic Patterns and Gateway Sizing

This section presents a high-level discussion of the differences between various traffic models or patterns and how they can affect voice gateway selection. The emphasis is on traffic patterns and gateway sizing for traffic-intensive deployments.

Definitions and Terminology

This section uses the following terms and definitions:

Simultaneous calls

The number of calls that are all active in the system at the same time.

Maximum simultaneous calls

The maximum number of simultaneous calls in active (talk) state that the system can handle. The number of calls expected to be active simultaneously during the busy hour of the day should not exceed this number.

Calls per second (cps)

The call arrival rate, described as the number of calls that arrive (that is, new call setup attempts) in one second. Call arrival rates are also often quoted in calls per hour, but this metric is looser in the sense that 100 calls arriving in the last five seconds of an hour provides an average call arrival rate of 100 calls per hour (which is an extremely low rate for a communications system), while it also provides an arrival rate of 20 calls per second (which is a high rate). Sustaining 20 calls per second for an entire hour would result in 72,000 calls per hour. Therefore calls-per-hour is not a very useful metric for ascertaining a system's ability to handle bursty call arrival traffic patterns.

Busy Hour Call Attempts (BHCA)

The number of calls attempted during the busiest hour of the day (the peak hour). This is the same as the calls-per-second rating for the busiest hour of the day, but it is expressed over a period of an hour rather than a second. For example, 10 cps would be equal to 36,000 calls per hour. There is also a metric for Busy Hour Call Completions (BHCC), which can be lower than the BHCA (call attempts) under the assumption that not all calls are successful (as when a blocking factor exists). This chapter assumes 100% call completions, so that BHCA = BHCC.

Bursty traffic

Steady arrival means the call attempts are spaced more or less equally over a period of time. For example, 60 calls per hour at a steady arrival rate would present one call attempt roughly every minute (or approximately 0.02 cps). With bursty arrival, the calls arriving over a given period of time (such as an hour) are not spaced equally but are clumped together in one or more spikes. In the worst case, an arrival rate of 60 calls per hour could offer all 60 calls in a single second of the hour, thus averaging 0 cps for most of the hour with a peak of 60 cps for that one second. This kind of traffic is extremely stressful to communications systems.

Hold time

This is the period of "talk time" on a voice call; that is, the period of time between call setup and call teardown when there is an open speech path between the two parties. A hold time of 3 minutes (180 seconds) is an industry average used for traffic engineering of voice systems. The shorter the hold time on the average call, the greater the percentage of system CPU time spent on setting up and tearing down calls compared to the CPU time spent on maintaining the speech path.

PSTN Traffic Patterns

Traffic, when used in the context of voice communication systems, refers to the volume of calls being sent and/or received. Of particular importance is the traffic carried by external circuits such as the public switched telephone network (PSTN). Traffic is measured in Erlangs, and an Erlang is defined as one call lasting for one hour. This section does not go into any further detail on Erlangs other than to say that there are mathematical tables (Erlang-B and Erlang-C) that are used to calculate how many circuits are required for a given amount of offered traffic.

The amount of traffic received and generated by your business determines the size of the external circuits required. However; many customers typically continue to use the same number of circuits for their IP-based communications system as they previously used for a TDM-based system. While this sizing method might work if no issues are encountered, the process of ongoing system traffic analysis should be part of any routine maintenance practices. Traffic analysis can show that the system is over-provisioned for the current levels of traffic (and, therefore, the customer is paying for circuits that are not needed) or, conversely, that the system is under-provisioned and may be suffering from occasional blocked and/or lost calls, in which case increasing the number of circuits will remedy the situation.

Normal Business Traffic Profile

Most customers have a normal traffic profile, which means that they typically have two busy hours per day, one occurring during the morning from 10:00 to 11:00 and the other in the afternoon from 14:00 to 15:00. These busy-hour patterns can often be attributed to such things as employees starting the work day or returning from a lunch break. The calls themselves tend to have longer hold times and they tend to arrive and leave in a steady manner. A generally accepted industry average holding time to use for traffic calculations is 3 minutes.

Assuming that the communications system is engineered with the busy-hour traffic in mind, no issues should arise. Engineering a system below these levels will result in blocked and/or lost calls, which can have a detrimental effect on business.

Contact Center Traffic Profile

Contact centers present somewhat different patterns of traffic in that these systems typically handle large volumes of calls for the given number of agents or interactive voice response (IVR) systems available to service them. Contact centers want to get the most out of their resources, therefore their agents, trunks, and IVR systems are kept busy all the while they are in operation, which usually is 24 hours a day. Call queuing is typical (when incoming call traffic exceeds agent capacity, calls wait in queue for the next available agent), and the agents are usually dedicated during their work shifts to taking contact center calls.

Call holding times in contact centers are often of a shorter average duration than normal business calls. Contributing to the shorter average call holding time is the fact that many calls interact only with the IVR system and never need to speak to a human agent (also termed self-service calls). A representative holding time for self-service calls is about 30 seconds, while a call that talks to an agent has an average holding time of 3 minutes (the same as normal business traffic), making the overall average holding time in the contact center shorter than for normal business traffic.

The goal of contact centers to optimize resource use (including IVR ports, PSTN trunks, and human agents), combined with the fact that contact centers are systems dedicated to taking telephone calls, also presents the system with higher call arrival rates than in a typical business environment. These call arrival rates can also peak at different times of day and for different reasons (not the usual busy hour) than normal business traffic. For example, when a television advertisement runs for a particular holiday package with a 1-800 number, the call arrival rate for the system where those calls are received will experience a peak of traffic for about 15 minutes after the ad airs. This call arrival rate can exceed the average call arrival rate of the contact center by an order of magnitude.

Gateway Sizing for Contact Center Traffic

Short call durations as well as bursty call arrival rates impact the PSTN gateway's ability to process the traffic. Under these circumstances the gateway needs more resources to process all calls in a timely manner, as compared to gateways that receive calls of longer duration that are presented more uniformly over time. Because gateways have varying capabilities to deal with these traffic patterns, careful consideration should be given to selecting the appropriate gateway for the environment in which it will operate. Some gateways support more T1/E1 ports than others, and some are more able than others to deal with multiple calls arriving at the same time.

For a traffic pattern with multiple calls arriving in close proximity to each other (that is, high or bursty call arrival rates), a gateway with a suitable rating of calls per second (cps) is the best fit. Under these conditions, using calls with 15-second hold times, the Cisco AS5400XM Universal Gateway can maintain 20 cps (with 310 calls active at once), the Cisco 3845 Integrated Services Router can maintain 17 cps (with 255 calls active at once), and the Cisco Catalyst 6500 Communication Media Module can maintain 7 cps (with 130 calls active at once). The performance of the Cisco AS5350XM Universal Gateway is identical to that of the AS5400XM in terms of calls per second.

For traffic patterns with a steady arrival rate, the maximum number of active calls that a gateway can handle is generally the more important consideration. Under these conditions, using calls with 180-second hold times, the Cisco AS5400XM Universal Gateway can maintain 630 simultaneously active calls (with a call arrival rate of up to 3.5 cps), the Cisco 3845 Integrated Services Router can maintain 504 simultaneously active calls (with a call arrival rate of up to 3 cps), and the Cisco Catalyst 6500 Communication Media Module can maintain 240 simultaneously active calls (with a call arrival rate of up to 1.3 cps).

These numbers assume that all of the following conditions apply:

CPU utilization does not exceed 75%.

PSTN gateway calls are made with ISDN PRI trunks using H.323.

Real Time Control Protocol (RTCP) timer is set to the default value of 5 seconds.

Voice Activity Detection (VAD) is off.

G.711 uses 20 ms packetization.

Cisco IOS Release 12.4.11T or later is used.

Dedicated voice gateway configurations are used, with ethernet (GE) egress and no QoS features. (Using QoS-enabled egress interfaces or non-ethernet egress interfaces, or both, will consume additional CPU resources.)

No supplementary call features or services are enabled - such as general security (for example, access control lists or firewalls), voice-specific security (TLS, IPSec and/or SRTP), AAA lookups, gatekeeper-assisted call setups, VoiceXML or TCL-enabled call flows, call admission control (RSVP), and SNMP polling/logging. Such extra call features will use additional CPU resources.

Voice Activity Detection (VAD)

VAD is a digital signal processing feature that suppresses the creation of most of the IP packets during times when the speech path in a particular direction of the call is perceived as being silent. Typically only one party on a call speaks at a time, so that packets need flow in only one direction, and packets in the reverse (or silent) direction need not be sent except as an occasional keepalive measure. VAD can therefore provide significant savings in the number of IP packets sent for a VoIP call, and thereby save considerable CPU cycles on the gateway platform. While the actual packet savings that VAD can provide varies with the call flow, the application, and the nature of speaker interactions, it tends to use 10% to 30% fewer packets than would be sent for a call made using a VAD-off configuration.

VAD is most often turned off in endpoints and voice gateways deployed in Unified CM networks; VAD is most often turned on in voice gateways in other types of network deployments.

Codec

Both G.711 and G.729A use as their default configuration a 20 ms sampling time, which results in a 50 packets per second (pps) VoIP call in each direction. While a G.711 IP packet (200 bytes) is larger than a G.729A packet (60 bytes), this difference has not proven to have any significant effect on voice gateway CPU performance. Both G.711 and G.729 packets qualify as "small" IP packets to the router, therefore the packet rate is the salient codec parameter affecting CPU performance.

Performance Overload

Cisco IOS is designed to have some amount of CPU left over during peak processing, to handle interrupt-level events. The performance figures in this section are designed with the processor running at an average load of approximately 75%. If the load on a given Cisco IOS gateway continually exceeds this threshold, the following will result:

The deployment will not be supported by Cisco Technical Assistance Center (TAC).

The Cisco IOS Gateway will display anomalous behavior, including Q.921 timeouts, longer post-dial delay, and potentially interface flaps.

Cisco IOS Gateways are designed to handle a short burst of calls, but continual overloading of the recommended call rate (calls per second) is not supported.


Note With any gateway, you might be tempted to assign unused hardware ports to other tasks, such as on a CMM gateway where traffic calculations have dictated that only a portion of the ports can be used for PSTN traffic. However, the remaining ports must remain unused, otherwise the CPU will be driven beyond supported levels.


Performance Tuning

The CPU utilization of a Cisco IOS Voice gateway is affected by every process that is enabled in a chassis. Some of the lowest level processes such as IP routing and memory defragmentation will occur even when there is no live traffic on the chassis.

Lowering the CPU utilization can help to increase the performance of a Cisco IOS Voice Gateway by ensuring that there are enough available CPU resources to process the real-time voice packets and the call setup instructions. Some of the techniques for decreasing CPU utilization are described in Table 4-2.

 

Table 4-2 Techniques for Reducing CPU Utilization 

Technique
CPU Savings
Description

Enable Voice Activity Detection (VAD)

Up to 20%

Enabling VAD can result in up to 45% fewer voice packets in typical conversations. The difficultly is that, in scenarios where voice recognition is used or there are long delays, a reduction in voice quality can occur. Voice appears to "pop" in at the beginning and "pop" out at the end of talk spurts.

Disable Real Time Control Protocol (RTCP)

Up to 5%

Disabling RTCP results in less out-of-band information being sent between the originating and terminating gateways. This results in lower quality of statistics displayed on the paired gateway. This can also result in the terminating gateway having a call "hang" for a longer period of time if RTCP packets are being used to determine if a call is no longer active.

Disable other non-essential functions such as: Authentication, Authorization, and Accounting (AAA); Simple Network Management Protocol (SNMP); and logging

Up to 2%

Any of these processes, when not required, can be disabled and will result in lower CPU utilization by freeing up the CPU to provide faster processing of real-time traffic.

Change call pattern to increase the length of the call (and reduce the number of calls per second)

Varies

This can be done by a variety of techniques such as including a long(er) introduction prompt played at the beginning of a call or adjusting the call script at the call center.


Additional Information

For more information on Cisco Voice Gateway capabilities and call center traffic analysis, refer to the following sources:

Cisco Voice Gateway Router Interoperability with Cisco Unified CallManager data sheet (Table 7):

http://www.cisco.com/en/US/prod/collateral/routers/ps259/product_data_sheet0900aecd8057f2e0.pdf

Cisco AS5400XM Universal Gateway data sheet (Table 9):

http://www.cisco.com/en/US/products/hw/univgate/ps505/products_data_sheet0900aecd802efc92.html

How to Order a Cisco AS5400XM Universal Gateway:

http://www.cisco.com/en/US/products/hw/univgate/ps505/prod_brochure0900aecd802f6ece.html

Various voice traffic calculators, including Erlang calculators:

http://www.erlang.com/calculator/

TDM and VoIP Trunking Gateways

Until approximately 2006, the only choice for an enterprise to connect its internal VoIP network to voice services outside the enterprise was via TDM gateways to the traditional PSTN. Cisco offers a full range of TDM gateways with analog and digital connections to the PSTN as well as to PBXs and key systems. TDM connectivity covers a wide variety of low-density analog (FXS and FXO), low density digital (BRI), and high-density digital (T1, E1, and T3) interface choices.

Starting around 2006, new voice service options to an enterprise started to become available from service providers, most often referred to as SIP trunk services. Using a SIP trunk for connecting to PSTN and other destinations outside the enterprise involves an IP-to-IP connection at the edge of the enterprise's VoIP network. The same functions traditionally fulfilled by a TDM gateway are still needed at this interconnect point, including demarcation, call admission control, ensuring QoS, a troubleshooting boundary, security checks, and so forth. For SIP trunking connections, the Cisco Unified Border Element fulfils these functions as a session border controller (SBC) at the interconnect point between the enterprise and the service provider network. Cisco Unified Border Element also performs protocol translation functions to interconnect H.323 and SIP equipment, or to interconnect SIP equipment using different variations of SIP implementations. Cisco Unified Border Element can also perform transcoding. If used for one of these functions, Cisco Unified Border Element may also be used internal to the enterprise network at interconnect points between equipment that cannot interoperate without a protocol translation or transcoding service.

TDM gateway platforms are discussed in detail in the remainder of this chapter. Cisco Unified Border Element is discussed in greater detail in the chapter on Cisco Unified CM Trunks. Both functions can be enabled on the same Cisco Integrated Services Router (ISR) platform at the same time.

Understanding Cisco Gateways

Cisco access gateways enable Cisco Unified Communications Manager (Unified CM) to communicate with non-IP telecommunications devices. There are two types of Cisco access gateways, analog and digital.

Cisco Access Analog Gateways

There are two categories of Cisco access analog gateways, trunk gateways and station gateways.

Access analog station gateways

Analog station gateways connect Unified CM to Plain Old Telephone Service (POTS) analog telephones, interactive voice response (IVR) systems, fax machines, and voice mail systems. Station gateways provide Foreign Exchange Station (FXS) ports.

Access analog trunk gateways

Analog trunk gateways connect Unified CM to PSTN central office (CO) or PBX trunks. Trunk gateways provide Foreign Exchange Office (FXO) ports for access to the PSTN, PBXs, or key systems, and E&M (recEive and transMit, or ear and mouth) ports for analog trunk connection to a legacy PBX. Whenever possible, use digital gateways to minimize any answer and disconnect supervision issues. Analog Direct Inward Dialing (DID) and Centralized Automatic Message Accounting (CAMA) are also available for PSTN connectivity.

Cisco Access Digital Trunk Gateways

A Cisco access digital trunk gateway connects Unified CM to the PSTN or to a PBX via digital trunks such as Primary Rate Interface (PRI), Basic Rate Interface (BRI), or T1 Channel Associated Signaling (CAS). Digital T1 PRI trunks may also be used to connect to certain legacy voice mail systems.

Tuning Gateway Gain Settings

Connecting a Cisco Unified Communications network to the PSTN through gateways requires that you properly address voice quality issues arising from echo and signal degradation due to power loss, impedance mismatches, delay, and so forth. For this purpose, you must establish a Network Transmission Loss Plan (NTLP), which provides a complete picture of signal loss in all expected voice paths. Using this plan, you can identify locations where signal strength must be adjusted for optimum loudness and effective echo cancellation. Note that not all carriers use the same loss plan, and that the presence of cellular networks adds further complexity in creating the NTLP. Cisco does not recommend adjusting input gain and output attenuation on gateways without first completing such an NTLP. For more information, refer to Echo Analysis for Voice Over IP, available at

http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/EA_ISD.pdf

Gateway Selection

When selecting an IP telephony gateway, consider the following factors:

Core Feature Requirements

Gateway Protocols

Gateway Protocols and Core Feature Requirements

Site-Specific Gateway Requirements

Core Feature Requirements

Gateways used in IP telephony applications must meet the following core feature requirements:

Dual tone multifrequency (DTMF) relay capabilities

DTMF relay capability, specifically out-of-band DTMF, separates DTMF digits from the voice stream and sends them as signaling indications through the gateway protocol (H.323, SCCP, MGCP, or SIP) signaling channel instead of as part of the voice stream or bearer traffic. Out-of-band DTMF is required when using a low bit-rate codec for voice compression because the potential exists for DTMF signal loss or distortion.

Supplementary services support

Supplementary services are typically basic telephony functions such as hold, transfer, and conferencing.

Fax/modem support

Fax over IP enables interoperability of traditional analog fax machines with IP telephony networks. The fax image is converted from an analog signal and is carried as digital data over the packet network. For more information, see Fax and Modem Support

Unified CM redundancy support

Cisco Unified Communications is based on a distributed model for high availability. Unified CM clusters provide for Unified CM redundancy. The gateways must support the ability to "re-home" to a secondary Unified CM in the event that a primary Unified CM fails. Redundancy differs from call survivability in the event of a Unified CM or network failure.

Refer to the gateway product documentation to verify that any IP Telephony gateway you select for an enterprise deployment can support the preceding core requirements. Additionally, every IP Telephony implementation has its own site-specific feature requirements, such as analog or digital access, DID, and capacity requirements (see Site-Specific Gateway Requirements).

Gateway Protocols

Cisco Unified CM (Release 3.1 and later) supports the following gateway protocols:

H.323

Media Gateway Control Protocol (MGCP)

Cisco Unified CM Release 4.0 and later supports Session Initiation Protocol (SIP) on the trunk side. The SIP trunk implementation has been enhanced in Cisco Unified CM releases 5.0 through 7.x to support more features.

Protocol selection depends on site-specific requirements and the installed base of equipment. For example, most remote branch locations have Cisco 2600XM, 2800, 3700, or 3800 Series routers installed. These routers support SIP, H.323, and MGCP 0.1 with Cisco IOS Release 12.2.11(T) and Cisco Unified CM Release 3.1 or later. For gateway configuration, MGCP might be preferred over H.323 or SIP due to simpler configuration. On the other hand, H.323 or SIP might be preferred over MGCP because of the robustness of the interfaces supported.

Simplified Message Desk Interface (SMDI) is a standard for integrating voice mail systems to PBXs or Centrex systems. Connecting to a voice mail system via SMDI and using either analog FXS or digital T1 PRI would require either SCCP or MGCP protocol because H.323 or SIP devices do not identify the specific line being used from a group of ports. Use of H.323 or SIP gateways for this purpose means the Cisco Message Interface cannot correctly correlate the SMDI information with the actual port or channel being used for an incoming call.

In addition, the Unified CM deployment model being used can influence gateway protocol selection. (Refer to the chapter on Unified Communications Deployment Models.)

Table 4-3 shows which gateways support a given protocol. Each of these protocols follows a slightly different methodology to provide support for the core gateway requirements. Gateway Protocols and Core Feature Requirements, describes how each protocol provides these feature requirements.

 

Table 4-3 Supported Gateway Protocols and Cisco Unified Communications Gateways 

Cisco Gateway
MGCP 0.1
H.323
SCCP
SIP

Cisco 2900

Yes, beginning with Cisco IOS Release 15.0.1M

Supported with:

FXS ports

Conferencing and transcoding DSP resources

Yes, beginning with Cisco IOS Release 15.0.1M

Supported with:

FXS ports

Conferencing and transcoding DSP resources

Yes, beginning with Cisco IOS Release 15.0.1M

Supported with:

FXS ports

Conferencing and transcoding DSP resources

Yes, SIP trunk

Cisco 3900

Yes, beginning with Cisco IOS Release 15.0.1M

Supported with:

FXS ports

Conferencing and transcoding DSP resources

Yes, beginning with Cisco IOS Release 15.0.1M

Supported with:

FXS ports

Conferencing and transcoding DSP resources

Yes, beginning with Cisco IOS Release 15.0.1M

Supported with:

FXS ports

Conferencing and transcoding DSP resources

Yes, SIP trunk

Cisco 3800

Yes, beginning with Cisco IOS Release 12.3.11T

Supported with:

Analog FXS/FXO

T1 CAS (E&M Wink Start; Delay Dial only)

T1/E1 PRI

Yes, beginning with Cisco IOS Release 12.3.11T

Yes for DSP resources, beginning with Cisco IOS Release 12.3.11T.

For FXS, use Cisco IOS Release 12.4.9.T or later.

Yes, SIP trunk

Cisco 2800

Yes, beginning with Cisco IOS Release 12.3.8T4

Supported with:

Analog FXS/FXO

T1 CAS (E&M Wink Start; Delay Dial only)

T1/E1 PRI

Yes, beginning with Cisco IOS Release 12.3.8T4

Yes for DSP resources, beginning with Cisco IOS Release 12.3.11T.

For FXS, use Cisco IOS Release 12.4.9.T or later.

Yes, SIP trunk

Cisco 3700

Yes

Supported with:

Analog FXS/FXO

T1 CAS (E&M Wink Start; Delay Dial only)

T1/E1 PRI

Yes

DSP farm in Cisco IOS Release 12.2.13T

Yes, SIP trunk

Communication Media Module (CMM)

Yes

Supported with:

T1 CAS FXS

T1/E1 PRI

FXS

Yes

No

Yes, SIP trunk

VGD-1T3

T3 with Cisco IOS Release 12.4.22T

Yes

Yes

Yes

VG224

Yes

FXS, conferencing, and transcoding.

Yes, FXS only

Yes, beginning with Cisco IOS Release 12.4(2)T

Yes, FXS only

VG248

No

No

Yes1

No

Cisco ATA 188

Yes, FXS only

Yes, FXS only

Yes, FXS only

Yes, third-party SIP phone

Cisco AS5350XM

Cisco AS5400XM

No

Yes

No

Yes, SIP trunk

VG202 and VG204

Yes, FXS only beginning with Cisco IOS Release 12.4(9th)T

Yes, FXS only beginning with Cisco IOS Release 12.4(9th)T

Yes, FXS only beginning with Cisco IOS Release 12.4(9th)T

Yes, FXS only beginning with Cisco IOS Release 12.4(9th)T

1 The VG248 is not a true gateway in that it uses Skinny Client Control Protocol (SCCP) instead of H.323, MGCP, or SIP.



Note Prior to deployment, check the Cisco IOS software release notes to confirm feature or interface support.


Gateway Protocols and Core Feature Requirements

This section describes how each protocol (SCCP, H.323, MGCP, and SIP) supports the following gateway feature requirements:

DTMF Relay

Supplementary Services

Unified CM Redundancy

DTMF Relay

Dual-Tone Multifrequency (DTMF) is a signaling method that uses specific pairs of frequencies within the voice band for signals. A 64 kbps pulse code modulation (PCM) voice channel can carry these signals without difficulty. However, when using a low bite-rate codec for voice compression, the potential exists for DTMF signal loss or distortion. An out-of-band signaling method for carrying DTMF tones across a Voice over IP (VoIP) infrastructure provides an elegant solution for these codec-induced symptoms.

SCCP Gateways

The Cisco VG248 carries DTMF signals out-of-band using Transmission Control Protocol (TCP) port 2002. Out-of-band DTMF is the default gateway configuration mode for the VG248.

H.323 Gateways

The H.323 gateways, such as the Cisco 3800 series products, can communicate with Unified CM using the enhanced H.245 capability for exchanging DTMF signals out-of-band. The following is an example out-of-band DTMF configuration on a Cisco IOS gateway:

dial-peer voice 100 voip
destination-pattern 555....
session target ipv4:10.1.1.1
CODEC g729ar8
dtmf-relay h245-alphanumeric
preference 0
 
   

MGCP Gateway

The Cisco IOS-based platforms use MGCP for Unified CM communication. (See Table 4-3 for a list of the Cisco gateway platforms that support MGCP.) Within the MGCP protocol is the concept of packages. The MGCP gateway loads the DTMF package upon start-up. The MGCP gateway sends symbols over the control channel to represent any DTMF tones it receives. Unified CM then interprets these signals and passes on the DTMF signals, out-of-band, to the signaling endpoint. The global configuration command for DTMF relay is:

mgcp dtmf-relay CODEC all mode out-of-band
 
   

You must enter additional configuration parameters in the Unified CM MGCP gateway configuration interface.

DTMF relay is enabled by default and does not need additional configuration.


Note Use the fm-package command to enable DTMF via RFC 2833, available as of Cisco IOS Release 12.4(6)T.


SIP Gateway

The Cisco IOS-based platforms can use SIP for Unified CM communication. (See Table 4-3 for a list of the Cisco gateway platforms that support SIP.) They support various methods for DTMF, but only the following methods can be used to communicate with Unified CM:

Named Telephony Events (NTE), or RFC 2833

Unsolicited SIP Notify (UN)

Key Press Markup Language (KPML)

The following example shows a configuration for NTE:

dial-peer voice 100 voip
destination-pattern 555....
session target ipv4:10.1.1.1
session protocol sipv2
dtmf-relay rtp-nte
 
   

The following example shows a configuration for UN:

dial-peer voice 100 voip
destination-pattern 555....
session target ipv4:10.1.1.1
session protocol sipv2
dtmf-relay sip-notify
 
   

For more details on DTMF method selection, see the chapter on Media Resources.

Supplementary Services

Supplementary services provide user functions such as hold, transfer, and conferencing. These are considered fundamental requirements of any voice installation. Each gateway evaluated for use in an IP telephony network should provide support for supplementary services natively, without the use of a software media termination point (MTP).

SCCP Gateways

The Cisco SCCP gateways provide full supplementary service support. They also support FXS SCCP ports with Cisco IOS Release 12.4.9T. The SCCP gateways use the Gateway-to-Unified CM signaling channel and SCCP to exchange call control parameters. (See Table 4-3 for a list of the Cisco gateway platforms that support SCCP.)

H.323 Gateways

H.323v2 implements Open/Close LogicalChannel and the emptyCapabilitySet features. The use of H.323v2 by H.323 gateways, beginning in Cisco IOS Release 12.0(7)T and Cisco Unified CM Release 3.0 and later, eliminates the requirement for an MTP to provide supplementary services. With Unified CM Release 3.1 and later, a transcoder is allocated dynamically only if required during a call to provide access to G.711-only devices while still maintaining a G.729 stream across the WAN. Full support for H.323v2 is available in Cisco IOS Release 12.1.1T.

Once an H.323v2 call is set up between a Cisco IOS gateway and an IP phone, using the Unified CM as an H.323 proxy, the IP phone can request to modify the bearer connection. Because the Real-Time Transport Protocol (RTP) stream is directly connected to the IP phone from the Cisco IOS gateway, a supported voice codec can be negotiated.

Figure 4-1 and the following steps illustrate a call transfer between two IP phones:

1. If IP Phone 1 wishes to transfer the call from the Cisco IOS gateway to Phone 2, it issues a transfer request to Unified CM using SCCP.

2. Unified CM translates this request into an H.323v2 CloseLogicalChannel request to the Cisco IOS gateway for the appropriate SessionID.

3. The Cisco IOS gateway closes the RTP channel to Phone 1.

4. Unified CM issues a request to Phone 2, using SCCP, to set up an RTP connection to the Cisco IOS gateway. At the same time, Unified CM issues an OpenLogicalChannel request to the Cisco IOS gateway with the new destination parameters, but using the same SessionID.

5. After the Cisco IOS gateway acknowledges the request, an RTP voice bearer channel is established between Phone 2 and the Cisco IOS gateway.

Figure 4-1 H.323 Gateway Supplementary Service Support

MGCP Gateway

The MGCP gateways provide full support for the hold, transfer, and conference features through the MGCP protocol. Because MGCP is a master/slave protocol with Unified CM controlling all session intelligence, Unified CM can easily manipulate MGCP gateway voice connections. If an IP telephony endpoint (for example, an IP phone) needs to modify the session (for example, transfer the call to another endpoint), the endpoint would notify Unified CM using SCCP. Unified CM then informs the MGCP gateway, using the MGCP User Datagram Protocol (UDP) control connection, to terminate the current RTP stream associated with the Session ID and to start a new media session with the new endpoint information. Figure 4-2 illustrates the protocols exchanged between the MGCP gateway, endpoints, and Unified CM.

Figure 4-2 MGCP Gateway Supplementary Service Support

SIP Gateway

The Unified CM SIP trunk interface to Cisco IOS SIP gateways supports supplementary services such as hold, blind transfer, and attended transfer. The support for supplementary services is achieved via SIP methods such as INVITE and REFER. For more details, refer to the following documentation:

Cisco Unified Communications Manager System Guide, available at

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html

Cisco IOS SIP Configuration Guide, available at

http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/12_4t/sip_12_4t_book.html

Unified CM Redundancy

An integral piece of the IP telephony architecture is the provisioning of low-cost, distributed PC-based systems to replace expensive and proprietary legacy PBX systems. This distributed design lends itself to the robust fault tolerant architecture of clustered Unified CMs. Even in its most simplistic form (a two-system cluster), a secondary Unified CM should be able to pick up control of all gateways initially managed by the primary Unified CM.

SCCP Gateways

Upon boot-up, the Cisco VG224, VG248, and ATA 188 gateways are provisioned with Unified CM server information. When these gateways initialize, a list of Unified CMs is downloaded to the gateways. This list is prioritized into a primary Unified CM and secondary Unified CM. In the event that the primary Unified CM becomes unreachable, the gateway registers with the secondary Unified CM.

H.323 VoIP Call Preservation for WAN Link Failures

H.323 VoIP call preservation enhancements for WAN link failures sustain connectivity for H.323 topologies where signaling is handled by an entity that is different from the other endpoint, such as a gatekeeper that provides routed signaling or a call agent (such as the Cisco BTS 10200 Softswitch, Cisco PGW2200 Softswitch, or Cisco Unified CM) that brokers signaling between the two connected parties. Call preservation is useful when a gateway and the other endpoint (typically a Cisco Unified IP Phone) are located at the same site but the call agent is remote and therefore more likely to experience connectivity failures.

H.323 call preservation covers the following types of failures and connections.

Failure Types:

WAN failures that include WAN links flapping or degraded WAN links.

Cisco Unified CM software failure, such as when the ccm.exe service crashes on a Unified CM server.

LAN connectivity failure, except when a failure occurs at the local branch.

Connection Types:

Calls between two Cisco Unified CM controlled endpoints under the following conditions:

During Unified CM reloads.

When a Transmission Control Protocol (TCP) connection used for signaling H.225.0 or H.245 messages between one or both endpoints and Unified CM is lost or flapping.

Between endpoints that are registered to different Unified CMs in a cluster, and the TCP connection between the two Unified CMs is lost.

Between IP phones and the PSTN at the same site.

Calls between a Cisco IOS gateway and an endpoint controlled by a softswitch, where the signaling (H.225.0, H.245 or both) flows between the gateway and the softswitch and media flows between the gateway and the endpoint:

When the softswitch reloads.

When the H.225.0 or H.245 TCP connection between the gateway and the softswitch is lost, and the softswitch does not clear the call on the endpoint.

When the H.225.0 or H.245 TCP connection between softswitch and the endpoint is lost, and the softswitch does not clear the call on the gateway.

Call flows involving a Cisco Unified Border Element (formerly, Cisco Multiservice IP-to-IP Gateway) running in media flow-around mode that reload or lose connection with the rest of the network.

Note that, after the media is preserved, the call is torn down later when either one of the parties hangs up or media inactivity is detected. In cases where there is a machine-generated media stream, such as music streaming from a media server, the media inactivity detection will not work and the call might hang. Cisco Unified CM addresses such conditions by indicating to the gateway that such calls should not be preserved, but third-party devices or the Cisco Unified Border Element would not do this.

Flapping is defined for this feature as the repeated and temporary loss of IP connectivity, which can be caused by WAN or LAN failures. H.323 VoIP calls between a Cisco IOS gateway and Cisco Unified CM may be torn down when flapping occurs. When Unified CM detects that the TCP connection is lost, it clears the call and closes the TCP sockets used for the call by sending a TCP FIN, without sending an H.225.0 Release Complete or H.245 End Session message. This is called quiet clearing. The TCP FIN sent from Unified CM could reach the gateway if the network comes up for a short duration, and the gateway will tear down the call. Even if the TCP FIN does not reach the gateway, the TCP keepalives sent from the gateway could reach Unified CM when the network comes up. Unified CM will send TCP RST messages in response to the keepalives because it has already closed the TCP connection. The gateway will tear down H.323 calls if it receives the RST message.

Configuration of H.323 VoIP call preservation enhancements for WAN link failures involves configuring the call preserve command. If you are using Cisco Unified Communications Manager, you must enable the Allow Peer to Preserve H.323 Calls parameter from the Service Parameters window.

The call preserve command causes the gateway to ignore socket closure or socket errors on H.225.0 or H.245 connections for active calls, thus allowing the socket to be closed without tearing down calls using those connections.

Example of H.323 VoIP Call Preservation for All Calls

The following configuration example enables H.323 VoIP call preservation for all calls:

voice service voip 
 h323 
  call preserve
 
   

MGCP Gateway

MGCP gateways also have the ability to fail over to a secondary Unified CM in the event of communication loss with the primary Unified CM. When the failover occurs, active calls are preserved.

Within the MGCP gateway configuration file, the primary Unified CM is identified using the call-agent <hostname> command, and a list of secondary Unified CM is added using the ccm-manager redundant-host command. Keepalives with the primary Unified CM are through the MGCP application-level keepalive mechanism, whereby the MGCP gateway sends an empty MGCP notify (NTFY) message to Unified CM and waits for an acknowledgement. Keepalive with the backup Unified CMs is through the TCP keepalive mechanism.

If the primary Unified CM becomes available at a later time, the MGCP gateway can "re-home," or switch back to the original Unified CM. This re-homing can occur either immediately, after a configurable amount of time, or only when all connected sessions have been released. This is enabled through the following global configuration commands:

ccm-manager redundant-host <hostname1 | ipaddress1 > <hostname2 | ipaddress2>
[no] call-manager redundancy switchback [immediate|graceful|delay <delay_time>]
 
   

SIP Gateway

Redundancy with Cisco IOS SIP gateways can be achieved similarly to H.323. If the SIP gateway cannot establish a connection to the primary Unified CM, it tries a second Unified CM defined under another dial-peer statement with a higher preference.

By default the Cisco IOS SIP gateway transmits the SIP INVITE request 6 times to the Unified CM IP address configured under the dial-peer. If the SIP gateway does not receive a response from that Unified CM, it will try to contact the Unified CM configured under the other dial-peer with a higher preference.

Cisco IOS SIP gateways wait for the SIP 100 response to an INVITE for a period of 500 ms. By default, it can take up to 3 seconds for the Cisco IOS SIP gateway to reach the backup Unified CM. You can change the SIP INVITE retry attempts under the sip-ua configuration by using the command retry invite <number>. You can also change the period that the Cisco IOS SIP gateway waits for a SIP 100 response to a SIP INVITE request by using the command timers trying <time> under the sip-ua configuration.

One other way to speed up the failover to the backup Unified CM is to configure the command monitor probe icmp-ping under the dial-peer statement. If Unified CM does not respond to an Internet Control Message Protocol (ICMP) echo message (ping), the dial-peer will be shut down. This command is useful only when the Unified CM is not reachable. ICMP echo messages are sent every 10 seconds.

The following commands enable you to configure Unified CM redundancy on a Cisco IOS SIP gateway:

sip-ua
 retry invite <number>
 timers trying <time>
 
   
dial-peer voice 101 voip
 destination-pattern 2...
 session target ipv4:10.1.1.101
 preference 0
 monitor probe icmp-ping
 session protocol sipv2
 
 
   
dial-peer voice 102 voip
 destination-pattern 2...
 session target ipv4:10.1.1.102
 preference 1
 monitor probe icmp-ping
 session protocol sipv2
 
   

Site-Specific Gateway Requirements

Each IP Telephony implementation has its own site-specific requirements. The following questions can help you with IP Telephony gateway selection:

Is the PSTN (or PBX) access analog or digital?

What type of analog (FXO, FXS, E&M, DID, CAMA) or digital (T1, E1, CAS, CCS) interface is required for the PSTN or PBX?

If the PSTN access is digital, what type of signaling is required (T1 CAS, Q.931 PRI, E1 CAS, or R2)?

What type of signaling does the PBX currently use?

FXO or FXS: loop start or ground start

E&M: wink-start, delay-start, or immediate-start

E&M: type I, II, III, IV, or V

T1: CAS, Q.931 PRI (User-Side or Network-Side), QSIG, DPNSS, or Proprietary d-channel (CCS) signaling

E1: CAS, R2, Q.931 PRI (User-Side or Network-Side), QSIG, DPNSS, Proprietary d-channel (CCS) signaling

What type of framing (SF, ESF, or G.704) and line encoding (B8ZS, AMI, CRC-4, or HDB3) does the PBX currently use?

Does the PBX require passing proprietary signaling? If so, which time slot is the signaling passed on, and is it HDLC-framed?

What is the required capacity of the gateway; that is, how many channels are required? (Typically, if 12 or more voice channels are required, then digital is more cost effective than an analog solution.)

Is Direct Inward Dialing (DID) required? If so, specify analog or digital.

Is Calling Line ID (CLID) needed?

Is Calling Name needed?

What types of fax and modem support are required?

What types of voice compression are required?

What types of supplementary services are required?

Will the PBX provide clocking, or will it expect the Cisco gateway to provide clocking?

Is rack space available for all needed gateways, routers, and switches?


Note Direct Inward Dial (DID) refers to a private branch exchange (PBX) or Centrex feature that permits external calls to be placed directly to a station line without use of an operator.



Note Calling Line Identification (CLI, CLID, or ANI) refers to a service available on digital phone networks to display the calling number to the called party. The central office equipment identifies the phone number of the caller, enabling information about the caller to be sent along with the call itself. CLID is synonymous with Automatic Number Identification (ANI).


Cisco Unified Communications gateways are able to inter-operate with most major PBX vendors, and they are EIA/TIA-464B compliant.

The site-specific and core gateway requirements are a good start to help narrow the possible choices. Once you have defined the required features, you can make a gateway selection for each of the pertinent configurations, whether they are single-site enterprise deployments of various sizes and complexities or multisite enterprise deployments.

The following tables summarize the features and interface types supported by the various Cisco gateway models.


Note In the following tables, the Cisco IOS and Unified CM release numbers refer to the minimum release that can support the listed feature on a particular gateway platform. For more information about Cisco IOS features, refer to the Cisco Feature Navigator located at http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp.


Cisco Analog Gateways

Table 4-4 lists supported interface types for Cisco analog gateways using H.323 or Session Initiation Protocol (SIP), and Table 4-5 lists supported interface types for Cisco analog gateways using Media Gateway Control Protocol (MGCP).

 

Table 4-4 Supported Analog H.323 and SIP Features 

Cisco Gateway
Interface Type
FXS
FXO
E&M
FXO, Battery Reversal
Analog DID
CAMA 911

3900 Series

Yes

Yes

Yes

Yes

Yes

Yes

2900 Series

Yes

Yes

Yes

Yes

Yes

Yes

VG202 and VG204

Yes

No

No

No

No

No

8800 Series

Yes

Yes

No

Yes

No

No

3800 Series

Yes

Yes

Yes

Yes

Yes

Yes

2800 Series

Yes

Yes

Yes

Yes

Yes

Yes

3700 Series1

Yes

Yes

Yes

Yes

Yes

Yes

Communication Media Module (CMM) 24FXS

Yes

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

N/A

N/A

N/A

N/A

N/A

N/A

VG224

Yes

N/A

N/A

N/A

N/A

N/A

VG248

No

No

No

No

No

No

Analog Telephone Adapter (ATA)1

Yes

No

No

No

No

No

7x00 family

N/A

N/A

N/A

N/A

N/A

N/A

1 These models have reached End of Sale (EOS).


 

Table 4-5 Supported Analog MGCP Features 

Cisco Gateway
Interface Type
FXS
FXO
E&M
FXO, Battery Reversal
Analog DID
CAMA 911

3900 Series

Yes

Yes

No

Yes

No

No

2900 Series

Yes

Yes

No

Yes

No

No

VG202 and VG204

Yes

No

No

No

No

No

8800 Series

Yes

Yes

No

Yes

No

No

3800 Series

Yes

Yes

No

Yes

No

No

2800 Series

Yes

Yes

No

Yes

No

No

3700 Series1

Yes

Yes

No

Yes

No

No

Communication Media Module (CMM) 24FXS

Yes

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

N/A

N/A

N/A

N/A

N/A

N/A

VG224

Yes

No

No

No

No

No

VG248

No

No

No

No

No

No

Analog Telephone Adapter (ATA)1

Yes

N/A

N/A

N/A

N/A

N/A

7x00 family

N/A

N/A

N/A

N/A

N/A

N/A

1 These models have reached End of Sale (EOS).


Cisco Digital Gateways

Table 4-6 through Table 4-9 list supported interface types for Cisco digital gateways using H.323 or Session Initiation Protocol (SIP). Table 4-10 lists supported interface types for Cisco digital gateways using Media Gateway Control Protocol (MGCP).

 

Table 4-6 Supported Digital H.323 and SIP Features for BRI, T1 CAS, T1 FGB, T1 FGD, and T1 QSIG 

Cisco Gateway
Interface Type
BRI (TE, User side)
BRI (NT, Network side)
BRI QSIG (Net3)
BRI Phones
T1 CAS (Robbed bit)
T1 FGB
T1 FGD
T1 QSIG

3900 Series

Yes

Yes

Yes

No

Yes

No

Yes

Yes

2900 Series

Yes

Yes

Yes

No

Yes

No

Yes

Yes

3800 Series

Yes

Yes

Yes

No

Yes

No

Yes

Yes

2800 Series

Yes

Yes

Yes

No

Yes

No

Yes

Yes

3700 Series1

Yes

Yes

Yes

No

Yes

No

Yes

Yes

5400XM

No

No

No

No

Yes

Yes

Yes

Yes

Communication Media Module (CMM) 24FXS

N/A

N/A

N/A

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

N/A

N/A

N/A

N/A

Yes

No

No

Yes

VG224

N/A

N/A

N/A

N/A

N/A

N/A

N/A

N/A

VG248

No

No

No

No

No

No

No

No

Analog Telephone Adapter (ATA)1

No

No

No

No

No

No

No

No

7x00 family

N/A

N/A

N/A

N/A

Yes

No

Yes

Yes

1 These models have reached End of Sale (EOS).


 

Table 4-7 Supported Digital H.323 and SIP Features for T1 PRI SL-1, 4ESS, and 5ESS 

Cisco Gateway
Interface Type
T1 PRI (User, DMS-100)
T1 PRI (Network, SL-1)
T1 PRI (User, 4ESS)
T1 PRI (Network, 4ESS)
T1 PRI (User, 5ESS)
T1 PRI (Network, 5ESS)

3900 Series

Yes

No

Yes

Yes

Yes

Yes

2900 Series

Yes

No

Yes

Yes

Yes

Yes

3800 Series

Yes

No

Yes

Yes

Yes

Yes

2800 Series

Yes

No

Yes

Yes

Yes

Yes

3700 Series1

Yes

No

Yes

No

Yes

No

5400XM

Yes

No

Yes

Yes

Yes

Yes

Communication Media Module (CMM) 24FXS

N/A

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

Yes

Yes

Yes

Yes

Yes

Yes

VG224

N/A

N/A

N/A

N/A

N/A

N/A

VG248

No

No

No

No

No

No

Analog Telephone Adapter (ATA)1

No

No

No

No

No

No

7x00 family

Yes

No

Yes

No

Yes

No

1 These models have reached End of Sale (EOS).


 

Table 4-8 Supported Digital H.323 and SIP Features for T1 PRI NI2, NFAS, and Network Specific Facilities (NSF) Service 

Cisco Gateway
Interface Type
T1 PRI (User, NI2)
T1 PRI (Network, NI2)
T1 PRI NFAS (User, DMS-100)
T1 PRI NFAS (User, 4ESS)
T1 PRI NFAS (User, 5ESS)
T1 PRI (Megacom or SDN, 4ESS)

3900 Series

Yes

Yes

Yes

Yes

Yes

Yes

2900 Series

Yes

Yes

Yes

Yes

Yes

Yes

3800 Series

Yes

Yes

Yes

Yes

Yes

Yes

2800 Series

Yes

Yes

Yes

Yes

Yes

Yes

3700 Series1

Yes

Yes

Yes

Yes

Yes

Yes

5400XM

Yes

Yes

Yes

Yes

Yes

Yes

Communication Media Module (CMM) 24FXS

N/A

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

Yes

Yes

Yes

Yes

Yes

No

VG224

N/A

N/A

N/A

N/A

N/A

N/A

VG248

No

No

No

No

No

No

Analog Telephone Adapter (ATA)1

No

No

No

No

No

No

7x00 family

Yes

Yes

No

No

No

No

1 These models have reached End of Sale (EOS).


 

Table 4-9 Supported Digital H.323 and SIP Features for E1 and J1 

Cisco Gateway
Interface Type
E1 CAS
E1 MELCAS
E1 R2
E1 PRI (User side, Net5)
E1 PRI (Network side, Net5)
E1 QSIG
J1 1

3900 Series

Yes

Yes

Yes

Yes

Yes

Yes

No

2900 Series

Yes

Yes

Yes

Yes

Yes

Yes

No

3800 Series

Yes

Yes

Yes

Yes

Yes

Yes

Yes

2800 Series

Yes

Yes

Yes

Yes

Yes

Yes

Yes

3700 Series2

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Communication Media Module (CMM) 24FXS

N/A

N/A

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

No

No

Yes

Yes

Yes

Yes

N/A

VG224

N/A

N/A

N/A

N/A

N/A

N/A

N/A

VG248

No

No

No

No

No

No

No

Analog Telephone Adapter (ATA)2

No

No

No

No

No

No

No

7x00 family

Yes

No

Yes

Yes

Yes

Yes

No

1 This interface type has reached End of Sale (EOS).

2 These models have reached End of Sale (EOS).


 

Table 4-10 Supported Digital MGCP Features 

Cisco Gateway
Interface Type
BRI 1
T1 CAS (E&M)
T1 CAS Hookflash
T1 PRI
T1 QSIG
E1 PRI
E1 QSIG
T3

3900 Series

15.0.1M

Yes2

Yes

Yes2

Yes2

Yes2

Yes2

No

2900 Series

15.0.1M

Yes2

Yes

Yes2

Yes2

Yes2

Yes2

No

3800 Series

12.4(2)T

Yes2

Yes

Yes2

Yes2

Yes2

Yes2

No

2800 Series

12.4(2)T

Yes2

Yes for 2811, 2821, 2851

Yes2

Yes2

Yes2

Yes2

No

3700 Series3

12.4(2)T

Yes2

Yes

Yes2

Yes2

Yes2

Yes2

No

VGD-1T3

No

No

No

No

No

No

No

Yes

Communication Media Module (CMM) 24FXS

N/A

N/A

Yes

N/A

N/A

N/A

N/A

No

CMM-6T1/E1

N/A

Yes

Yes

Yes

Yes

Yes

Yes

No

VG224

N/A

N/A

N/A

N/A

N/A

N/A

N/A

No

VG248

N/A

N/A

N/A

N/A

N/A

N/A

N/A

No

Analog Telephone Adapter (ATA)3

N/A

N/A

N/A

N/A

N/A

N/A

N/A

No

7x00 family

N/A

No

N/A

No

No

No

No

No

1 Cisco IOS Release 12.4(2)T supports BRI MGCP with the following hardware: NM-HDV2, NM-HD-XX and on-board H-WIC slots. BRI MGCP is also supported on older Cisco IOS releases with NM-1V/2V hardware.

2 AIM-VOICE-30 modules require Cisco IOS Release 12.2.13T to support MGCP.

3 These models have reached End of Sale (EOS).


QSIG Support

QSIG is a suite of international standards designed to provide flexibility in connecting PBX equipment to a corporate network. Among its other features, QSIG provides an open, standards-based method for interconnecting PBX equipment from different vendors.

ECMA QSIG is currently supported in H.323 gateways in PBX-to-PBX mode. The H.323 gateways provide full QSIG feature transparency for QSIG information elements. Basic call setup and teardown are supported using H.323 QSIG gateways, as summarized in Table 4-11.

 

Table 4-11 QSIG Support on H.323 Gateways 

Platform
Media
Minimum Cisco IOS Software Release Required

Cisco 3900 Series

BRI and T1/E1 QSIG

15.0.1M

Cisco 2900 Series

BRI and T1/E1 QSIG

15.0.1M

Cisco 3800 Series

BRI and T1/E1 QSIG

12.3.11T

Cisco 2800 Series

BRI and T1/E1 QSIG

12.3.8T4

Cisco 37001

T1/E1 QSIG

12.2.8T

Cisco AS5350XM

Cisco AS5400XM

T1/E1

12.2.2T

1 These models have reached End of Sale (EOS).


For more information on QSIG support on Cisco IOS gateways, refer to the documentation at

http://www.cisco.com/en/US/docs/ios/12_1t/12_1t2/feature/guide/dt_qsig.html

Prior to Cisco Unified CM Release 3.3, basic PRI functionality is all that is provided whenever a PBX is connected to a gateway using QSIG via H.323 and calls are made between phones on the PBX and IP phones attached to the Unified CM. This basic functionality, which includes only the Calling Line Identifier (CLID) and Direct Inward Dialed (DID) number, is provided by the gateway terminating the QSIG protocol rather than by Unified CM.

For Unified CM to support QSIG functionality, QSIG must be back-hauled directly to Unified CM. This support is provided in Cisco Unified CM 3.3 and later releases, in conjunction with all MGCP T1/E1 ISDN gateways.

Fax and Modem Support

This section describes the fax and modem support available with Unified CM and Cisco voice gateways. This section first presents brief overviews of fax and modem support on Cisco voice gateways, followed by a listing of supported platforms and example configuration files.

Gateway Support for Fax Passthrough and Fax Relay

Fax over IP enables interoperability of traditional analog fax machines with IP Telephony networks. The fax image is converted from an analog signal and is carried digitally over the packet network.

In its original form, fax data is digital and is contained in High-Level Data Link Control (HDLC) frames. However, to transmit across a traditional PSTN, these digital HDLC frames are modulated onto an analog carrier. While this analog carrier is necessary for effective faxing in PSTN environments, it is not ideal for the type of digital transport used by IP packet networks. Therefore, specific transport methods have been devised for successful transport of fax transmissions over an IP infrastructure.

The two main methods for transporting fax over IP are passthrough and relay. Passthrough is the simplest method, and it works by sampling and digitizing the analog fax signal just like a voice codec does for human speech. While there are a number of codecs available, passthrough always uses the G.711 codec for carrying fax information because it offers the least distortion of the analog fax signals. If a high-compression codec is being used by the original voice call, then passthrough uses an upspeed feature to change the codec to G.711. Passthrough is also commonly referred to as Voice Band Data (VBD), and Cisco provides two versions of passthrough: modem passthrough and fax passthrough.

Modem passthrough uses Cisco proprietary Named Signaling Event (NSE) packets to switch the call from voice mode to passthrough mode. This switchover from voice mode to passthrough is an important concept for passthrough and relay as well. Every call on a Cisco voice gateway starts out as a voice call, and the proper switchover occurs only when the gateway determines that the call is truly a fax call. While modem passthrough uses NSE packets for its switchover, other fax and modem transport methods may implement different mechanisms.

Despite its name, modem passthrough is also widely used for fax calls. Modem passthrough is also referred to as NSE-based passthrough, and you can activate it in the Cisco IOS command line interface (CLI) by using the modem passthrough command.

Fax passthrough is also referred to as protocol-based passthrough because it relies on the underlying call control protocol to switch the call from voice mode to passthrough. Fax passthrough works only with the call control protocols of H.323 and SIP. Because fax passthrough utilizes the call control protocol for its switchover, this is the only passthrough solution that can work with third party devices.

Figure 4-3 highlights the two different passthrough implementations employed by Cisco voice gateways for fax calls.

Figure 4-3 Cisco Passthrough Implementations for Fax Calls

Relay is the other main method for transporting fax over IP, and its implementation is a bit more complicated than passthrough. Relay strips off the analog carrier from the fax signal in a process known as demodulation to expose the fax HDLC data frames. The pertinent information in these HDLC frames is then removed and efficiently packaged in a fax relay protocol to be transported to the gateway on the other side. Once received on the other side, the fax information is pulled from the relay protocol, reconstructed back into fax HDLC frames, and modulated onto an analog carrier for transmission to a fax machine.

Cisco supports two versions of fax relay, T.38 and Cisco fax relay. An ITU standard, T.38 allows Cisco gateways to interoperate with third-party devices that also support the T.38 specification. In most scenarios, T.38 fax relay uses the call control protocol to switch from voice mode to T.38 fax relay mode, and this is referred to as protocol-based T.38 fax relay. However, it is also possible to configure T.38 fax relay to switch over using Cisco proprietary NSEs in what is termed NSE-based T.38 fax relay. In order to ensure third-party interoperability, protocol-based T.38 must be utilized.

Cisco fax relay is a pre-standard implementation, and it is proprietary to Cisco voice gateways. It is also the default fax transport configuration on nearly all Cisco voice gateways. Unlike the NSE or protocol-based methods used by T.38 fax relay and passthrough, Cisco fax relay transitions from voice to relay mode utilizing specific RTP dynamic payload types (PT). Figure 4-4 illustrates the Cisco fax relay methods.

Figure 4-4 Cisco Relay Implementations for Fax Calls

Fax relay mode, and more specifically T.38, is the preferred method to transport fax traffic. However, if T.38 fax relay is not supported, then Cisco fax relay or passthrough can be used as an alternative.

Best Practices

The following recommendations and guidelines can assist you in best implementing fax support on Cisco voice gateways:

When using QoS, make every effort to minimize the following:

Packet loss

Delay

Delay variation (jitter)

Fax transmissions are extremely sensitive to packet loss. Even minimal packet loss can cause fax failures. If packet loss is a problem in your network, then the redundancy feature in T.38 fax relay should be used. Also, ensure that constant packet delay on the network does not exceed 1 second and that delay variation (jitter) does not exceed 300 milliseconds. For detailed information about implementing QoS in a Cisco Unified Communications network, refer to the Enterprise QoS Solution Reference Network Design Guide, available at

http://www.cisco.com/go/designzone

The following tips can help ensure the integrity of the fax calls:

Use call admission control (CAC) to ensure that calls are not admitted if they exceed the specified total bandwidth limit.

Disable call waiting on all dedicated modem and fax ports.

T.38 fax relay provides the best fax performance based on network considerations and is the recommended transport method for fax traffic.

To insure interoperability with other vendor's T.38 products, use protocol-based T.38.

NSE-based T.38 must be used for communicating with certain Cisco voice gateways, such as the Cisco VG248 and any Cisco IOS SCCP gateways.

In Unified CM scenarios where T.38 is to be deployed among gateways running a variety of call signaling protocols, protocol-based T.38 should be the first choice. Beginning with Cisco Unified CM Release 6.0, Unified CM supports protocol-based T.38 with H.323, SIP, and MGCP call control protocols. If protocol-based T.38 is not supported in your installed version of Cisco Unified CM or if SCCP gateways are involved, then NSE-based T.38 should be used. To verify if your version of Unified CM supports protocol-based T.38, refer to the Cisco Unified Communications Manager release notes available at

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.html

T.38 fax relay is supported on most of the current Cisco voice gateways, especially those running Cisco IOS. Notable exceptions are the Cisco Analog Telephone Adaptors (ATAs) and legacy products such as the Cisco 6608, 6624, and DT-24/DE-30+.

Most Cisco voice gateways support Cisco fax relay, and it is the default fax transport method. Notable exceptions with regard to the support of Cisco fax relay are the Cisco AS5350 and AS5400 with Nextport DSP cards, the Cisco ATA, and the Cisco DT-24/DE-30+. Another exception is that the Cisco 2900 and 3900 Series gateways with the PVDM3 DSPs do not support Cisco fax relay.

Unlike T.38 fax relay, which uses an Unnumbered Datagram Protocol Transport Layer (UDPTL) header, Cisco fax relay utilizes the standard RTP header, which makes Cisco fax relay capable of secure fax transactions using Secure Real-Time Transport Protocol (SRTP).

Modem passthrough is supported by all current Cisco voice gateways, including the Cisco ATA as well as most legacy voice gateways.

Because modem passthrough uses a Cisco proprietary NSE-based switchover, it is not compatible with other vendor's equipment. However, fax passthrough with its protocol-based switchover should interoperate with most third-party devices if a passthrough solution is necessary.

Fax passthrough is supported only on Cisco IOS voice gateways that use the H.323 and SIP protocols.

Most fax machines appear to accept packet drop in the range of 0.4% to 0.6% without slowing down to the next speed. However, in a network with packet drop in the range of 0.8% to 1%, you should disable Error Correction Mode (ECM).

You can disable ECM on the gateway itself rather than disabling it on multiple fax machines. However, if packet drops occur, the fax image quality might deteriorate. Therefore, you should disable ECM only after considering whether you want to risk compromising image quality rather than experiencing longer call durations or dropped calls. You should also monitor and evaluate the network to identify and resolve the cause of the dropped packets.

Super-Group 3 Fax Support

Cisco IOS gateways with Cisco IOS Release 12.4.4T support Super-Group 3 (SG3) fax; however, only Group 3 speeds are negotiated. Beginning with Cisco IOS Release 15.0.1M, SG3 fax is supported natively. For more information on this feature, refer to Fax Relay Support for SG3 Fax Machines at G3 Speeds, available at

http://www.cisco.com/en/US/products/ps6441/products_feature_guide09186a00805138e5.html

If it is necessary to transport SG3 high-speed faxes at their native speeds, then modem passthrough must be used.

Gateway Support for Modem Passthrough and Modem Relay

In general, there are three mechanisms for supporting modem sessions over an IP network using voice gateways:

Modem passthrough

Cisco Modem Relay

Secure Modem Relay (Secure Communication Between STE Endpoints)

Each of these mechanisms can transport modem calls, but the relay methods are restrictive in that only certain modem modulations are supported. Modem passthrough, on the other hand, can handle any modulation.

An important concept to understand when dealing with the transport of modem signals across IP networks is the switchover that must occur on the gateway. Every call on a Cisco gateway begins as a voice call initially. Even if the call is between modems, the call will be set up as a voice call first. Then, once the gateway is sure that the call is truly a modem call, a switchover occurs that converts the gateway from voice call mode to a modem passthrough or modem relay mode. There are various switchover methods to transition a call from voice mode to modem passthrough or relay.

As discussed previously in the section on Gateway Support for Fax Passthrough and Fax Relay, modem passthrough uses proprietary NSE packets to switch a voice call into passthrough mode. When modem signals are detected, the gateways use these NSE messages to inform each other of the impending modem call. The gateways then make adjustments to better handle the transport of the modem signals. These adjustments include upspeeding the voice codec to G.711, disabling Voice Activity Detection (VAD), and disabling the echo cancellers if necessary. Because modem passthrough simply samples the analog modem signal using the G.711 codec, it should handle any modem modulation, but not always at the highest speeds.

Cisco Modem Relay is a proprietary implementation that efficiently transports V.34 modem calls over an IP network. V.90 calls are also supported, but they are forced to train down to V.34 speeds. As with modem passthrough, NSE packets are used to handle the switchover to Cisco Modem Relay from voice mode.

Secure modem relay, which is also referred to as Secure Communication Between STE Endpoints, allows for the transport of secure telephone calls over an IP infrastructure. Special devices known as Secure Terminal Equipment (STE) transmit encrypted voice using the V.32 modulation. Secure modem relay is designed to handle the transport of information between STEs in Unified CM environments with SCCP and MGCP gateways. Secure modem relay is not compatible with Cisco Modem Relay. One of the main reasons is that the switchover for secure modem relay does not use NSEs but instead uses V.150.1-based State Signaling Event (SSE) messages.

Secure modem relay is designed specifically for transporting STE signals and is almost never used outside of government or defense-related deployments. In most cases, Cisco Modem Relay or modem passthrough should be used for transporting modem calls. For more information on secure modem relay, refer to Secure Communication Between IP-STE Endpoint and Line-Side STE Endpoint, available at

http://www.cisco.com/en/US/products/ps6441/products_feature_guide09186a0080513c09.html

Figure 4-5 summarizes the Cisco modem transport implementations. Modem relay should be used whenever possible because it offers the most bandwidth efficiency and tolerance for network impairments when compared to modem passthrough. The disadvantage of modem relay is that it is quite restrictive on the modulations supported, while modem passthrough can handle any modem modulation.

Figure 4-5 Cisco Passthrough and Relay Implementations for Modem Calls

Best Practices

Observe the following recommended best practices to ensure optimum performance of modem traffic transported over an IP infrastructure:

Ensure that the IP network is enable for Quality of Service (QoS) and that you adhere to all of the recommendations for providing QoS in the LAN, MAN, and WAN environments. Every effort should be made to minimize the following parameters:

Packet loss — Fax and modem traffic requires an essentially loss-free transport. A single lost packet will result in retransmissions.

Delay

Delay variation (jitter)

For more information, refer to the Enterprise QoS Solution Reference Network Design Guide, available at

http://www.cisco.com/go/designzone

Use call admission control (CAC) to ensure that calls are not admitted if they exceed the specified total bandwidth limit.

Use modem relay whenever possible. Modem passthrough should be used for any modulations not supported by modem relay.

Do not use the IP network to connect modems that will be used to troubleshoot or diagnose problems with the IP network. In this case, the modems used to troubleshoot the devices that compose the IP infrastructure should be connected to a plain old telephone service (POTS).

Because of the NSE switchover utilized by Cisco modem relay and modem passthrough, gateways using different call control protocols can easily communicate with one another. For example, an MGCP gateway and an H.323 gateway connected to Unified CM can successfully negotiate Cisco modem relay or modem passthrough because the NSE switchover occurs within the RTP voice media stream that has already been set up by Unified CM.

Disable call waiting on all dedicated modem and fax ports.

V.90 Support

Currently, Cisco equipment supports only V.34 modems. Although V.90 modems will function on existing hardware, and speeds higher than V.34 speeds can be achieved, full V.90 support cannot be guaranteed.

Supported Platforms and Features

The following Cisco platforms support fax and modem features:

Cisco IOS Gateways support:

Modem passthrough

Fax passthrough for the H.323 and SIP protocols

T.38 fax relay. Both NSE and protocol-based switchovers for T.38 are supported, except in the case of SCCP where only NSE-based T.38 fax relay is supported.

Cisco fax relay. The Cisco AS5350, AS5400, and AS5850 using Nextport DSP cards do not support Cisco fax relay, and neither do the Cisco 2900 and 3900 Series using PVDM3 DSPs.

Cisco modem relay

Cisco non-IOS gateways:

The Cisco VG248 supports modem passthrough, NSE-based T.38 fax relay, and Cisco fax relay.

The Cisco 6608 and 6624 support only modem passthrough and Cisco fax relay.

The Cisco ATAs support modem passthrough for fax calls only. Using modem passthrough with an ATA for modem calls is not officially supported.


Note The fax and modem support information presented here is valid beginning with Cisco IOS Release 12.4(9)T for the Cisco IOS gateways and Release 1.3.1 of the Cisco VG248 Analog Phone Gateway.


Platform Protocol Support

Common call control protocols used today in enterprise solutions include H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP). Not all Cisco voice platforms support all of these protocols or all of the fax and modem features, thus raising interoperability issues. Additional interoperability issues occur when mixing Cisco IOS gateways, such as the Cisco 2800 Series or the Cisco 3800 Series, with non-IOS gateways such as the VG248. This section lists the combinations of gateways that provide support for interoperability of fax, modem, and protocol features.

Some of the common combinations of protocols in a network include: MGCP and H.323; SCCP and H.323; SCCP and SIP; MGCP and SIP; H.323 and SIP; and SCCP and MGCP. Common voice gateways included the Cisco VG224, VG248, 2600XM, 2800, 3700, 3800, and Catalyst 6000.

Table 4-12 lists the protocol combinations that currently support fax and modem interoperability.

 

Table 4-12 Fax and Modem Features Supported with Various Combinations of Call Control Protocols 

Protocol Combinations
Modem Relay
Modem Passthrough 1
T.38 Fax Relay
Cisco Fax Relay
Fax Passthrough

Unified CM using MGCP combined with Unified CM using H.323 or SIP

Yes

Yes

Yes2

Yes

Yes

Unified CM using MGCP combined with Unified CM using MGCP

Yes

Yes

Yes2

Yes

Yes

SCCP combined with Unified CM using H.323 or SIP

Yes

Yes

Yes3

Yes

Yes

SCCP combined with Unified CM using MGCP

Yes

Yes

Yes3

Yes

Yes

Unified CM using H.323 combined with H.323 or SIP

Yes

Yes

Yes2

Yes

Yes

Unified CM using SIP combined with H.323 or SIP

Yes

Yes

Yes2

Yes

Yes

1 Modem passthrough works for both modem and fax passthrough calls.

2 NSE-based T.38 fax relay works, but protocol-based T.38 fax relay depends on the version of Unified CM. For version information, refer to the Cisco Unified Communications Manager release notes available at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.html.

3 SCCP protocol works only with NSE-based T.38 fax relay.



Note Table 4-12 is a general reference. You should be aware that specific products might have limitations that are not listed in this table. For example, the Cisco ATA supports H.323, SIP, and SCCP call control protocols, but only modem passthrough is supported no matter which call control protocol is used.


Gateway Configuration Examples

This section provides a quick configuration overview for fax and modem support on Cisco gateways. More detailed configuration information can be found in the Cisco IOS Fax and Modem Services over IP Application Guide, which is available to Cisco employees and partners (with appropriate login authentication) at

http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps5207/products_configuration_guide_book09186a0080762024.html

Cisco fax relay is enabled by default on all voice gateways that support it. It must be disabled explicitly or it will attempt to transport any fax call that is detected by the voice gateway. If a feature such as modem passthrough is going to be used to transport fax calls, then Cisco fax relay can be disabled on Cisco IOS gateways by configuring fax protocol none under the dial-peer or globally under voice service voip. In many cases, however, it might be optimal to keep Cisco fax relay enabled to handle fax calls while configuring modem passthrough to handle high-speed modem calls and SG3 fax calls.

Cisco IOS Gateway Configuration for Modem Passthrough

Modem passthrough can be enabled for H.323 and MGCP gateways as shown in the following examples. SIP gateways use the same command as shown in the H.323 example. Also, for both H.323 and SIP voice gateways, modem passthrough can be enabled globally for all dial peers under voice service voip.

H.323

!
! Cisco fax relay is ON by default
!(except for 5350/5400, where Cisco fax relay is not supported)
!
dial-peer voice 1000 voip
 destination-pattern 1T
 session target ipv4:10.10.10.1
 modem passthrough mode nse codec g711ulaw
!
!
 
   

MGCP

!
ccm-manager mgcp
mgcp
mgcp call-agent 10.10.10.1 service-type mgcp version 0.1
mgcp modem passthrough voip mode nse
mgcp fax t38 inhibit
!
dial-peer voice 100 pots
 application mgcpapp
 port 1/0/0
!
 
   

Cisco VG248 Configuration for Modem Passthrough

The Cisco VG248 also handles both Cisco fax relay and modem passthrough, with Cisco fax relay being enabled by default. Cisco fax relay can be enabled or disabled from the "Port specific parameters" section of the VG248's telephone configuration.

When configuring modem passthrough on the VG248, two important parameters need to be configured. First, under Port specific parameters, the Passthrough mode must be set to default: automatic. Second, under the Telephony Advanced settings, the Passthrough signalling must be set to IOS mode, as shown below.

  -----------------------------------------------------------
  |                Cisco VG248 (VGC10d8002407)               |
  ----- -----------------------------------------------------
       | Advanced settings                                   |
       |-----------------------------------------------------|
       | Allow last good configuration (enabled)             |
       | SRST policy                   (disabled)            |
       | SRST provider                 ()                    |
       | Call preservation             (enabled: no timeout) |
       | Media receive timeout         (disabled)            |
       | Busy out off hook ports       (disabled)            |
       | DTMF tone dur ------------------------  100ms)      |
       | Echo cancelli| Passthrough signalling |e: use DSP)  |
       | Passthrough s|------------------------|)            |
       | Hook flash ti| legacy                 | default>)   |
       | Hook flash re| IOS mode               |             |
       | Fax relay max ------------------------  14400 bps)  |
       | Fax relay playout delay       (default: 300)        |
        -----------------------------------------------------
          -------------------------------------------------
 
   
  -----------------------------------------------------------
  |                Cisco VG248 (VGC10d8002407)               |
  ----- -----------------------------------------------------
       | Advanced settings                                   |
       |-----------------------------------------------------|
       | Allow last good configuration (enabled)             |
       | SRST policy                   (disabled)            |
       | SRST provider                 ()                    |
       | Call preservation             (enabled: no timeout) |
       | Media receive timeout         (disabled)            |
       | Busy out off hook ports       (disabled)            |
       | DTMF tone duration            (default: 100ms)      |
       | Echo cancelling policy        (alternate: use DSP)  |
       | Passthrough signalling        (IOS mode)            |
       | Hook flash timer              (<country default>)   |
       | Hook flash reject period      (none)                |
       | Fax relay maximum speed       (default: 14400 bps)  |
       | Fax relay playout delay       (default: 300)        |
        -----------------------------------------------------
         -------------------------------------------------
 
   

Clock Sourcing for Fax and Modem Passthrough

The clock signal plays a critical role in allowing fax and modem transmissions to work correctly. The gateway clock must synchronize with the PSTN clock, where Stratum clocking is provided. Without this clock synchronization, fax and modem communications will not work. To synchronize the clocks correctly, enter the following configuration for the T1 controller in a Cisco IOS gateway. (In this example, the T1 controller is the voice gateway that connects to the PSTN.)

!
controller T1 0
 framing esf
 linecode b8zs
 clock source line
 channel-group 1 timeslots 1-24 speed 64
 !
 
   

Also enter this configuration in all other interfaces connected to the PSTN.

T.38 Fax Relay

T.38 fax relay is not supported on Cisco ATA 188, 6608, and 6624 gateways, but it is supported on the Cisco IOS voice platforms and the VG248.

You can configure T.38 fax relay in the following ways:

Named Service Event (NSE) T.38 Fax Relay

Protocol-Based T.38 Fax Relay

Named Service Event (NSE) T.38 Fax Relay

The configuration of NSE-based T.38 fax relay for H.323 and SIP can be accomplished at the dial-peer level or globally under voice service voip for Cisco IOS gateways. The following example is for an H.323 dial-peer configuration, but the same command syntax is applicable to SIP dial peers as well.

H.323

!
dial-peer voice 1000 voip
 destination-pattern 1T
 session target ipv4:10.10.10.1
 modem passthrough mode nse codec g711ulaw
 fax protocol t38 nse
!
!
 
   

MGCP

For Cisco IOS MGCP gateways, NSE-based T.38 fax relay is often referred to as gateway controlled T.38 mode because the gateways control the T.38 switchover through the NSE messages. Gateway-controlled T.38 fax relay is enabled with the command no mgcp fax t38 inhibit.

!
ccm-manage mgcp
mgcp
mgcp call-agent 10.10.10.1 service-type mgcp version 0.1
mgcp modem passthrough voip mode nse
no mgcp fax t38 inhibit
!
dial-peer voice 100 pots
 application mgcpapp
 port 1/0/0
!
!
 
   

For SCCP gateways (either Cisco IOS gateways or the VG248), NSE-based T.38 fax relay must be used. Make sure that the command fax protocol t38 nse is configured under voice service voip if you want NSE-based T.38 fax relay to work on Cisco IOS SCCP gateways.

In situations where gateways are using different call control protocols, the NSE switchover might have to be "forced". When the same call control protocol is being used, the gateways will notify one another during the call setup that NSE-based T.38 fax relay is supported. When different call protocols are used, such as one gateway using H.323 and the other gateway using MGCP, this NSE-based T.38 fax relay verification might not get passed between the gateways. Therefore, the gateways must be programmed to force the NSE negotiation even though a notification of NSE-based T.38 support has not been received. For H.323 and SIP voice gateways, this is simply accomplished by adding the force option to the existing t38 configuration command fax protocol t38 nse force. For MGCP, use the command mgcp fax t38 gateway force.

Protocol-Based T.38 Fax Relay

With protocol-based T.38 fax relay, the switchover to T.38 from voice mode occurs within the call control protocol. The call control protocols supported by protocol-based T.38 fax relay are H.323, SIP, and MGCP. For H.323 and SIP, protocol-based T.38 can be configured at the dial peer level or globally under voice service voip. The command syntax is the same as for NSE-based T.38 fax relay, except that the nse keyword is omitted.

An additional fallback option may also be specified for both NSE-based and protocol-based T.38 fax relay for the H.323 and SIP call control protocols. This option allows the gateway to attempt another switchover method or even an entirely different transport method if the initial fax transport selection fails to negotiate. The following example illustrates the configuration of protocol-based T.38 fax relay along with the fallback option for an H.323 dial peer. The command syntax is the same for SIP dial peers and for global configurations under voice service voip.

H.323

!
dial-peer voice 1000 voip
 destination-pattern 1T
 session target ipv4:10.10.10.1
 modem passthrough mode nse codec g711ulaw
!
! To enable T.38 fax relay and fall back to Cisco fax relay when
! T.38 fax negotiation fails. This is the default case.
 fax protocol t38 fallback cisco
!
dial-peer voice 1001 voip
 destination-pattern 2T
 session target ipv4:10.10.10.2
 modem passthrough mode nse codec g711ulaw
!
! To enable T.38 fax relay and fall back to fax passthrough when
! T.38 fax negotiation fails.
fax protocol t38 nse fallback pass-through
!
dial-peer voice 1002 voip
 destination-pattern 3T
 session target ipv4:10.10.10.3
 modem passthrough mode nse codec g711ulaw
!
! This CLI is needed when talking to MGCP endpoint where CA/GK 
! doesn't support T.38 fax relay such as CCM.
 fax protocol t38 nse force fallback none
!
!
 
   

MGCP

Protocol-based T.38 fax relay for MGCP voice gateways is commonly referred to as CA-controlled T.38 mode because the call agent (CA) handles the T.38 fax relay switchover. As shown in the following examples, you must make sure that T.38 fax relay is enabled for MGCP and that the two fxr-package commands are also configured.

!
ccm-manage mgcp
mgcp
mgcp call-agent 10.10.10.1 service-type mgcp version 0.1
mgcp modem passthrough voip mode nse
no mgcp fax t38 inhibit
mgcp package-capability fxr-package
mgcp default-package fxr-package
!
dial-peer voice 100 pots
 application mgcpapp
 port 1/0/0
!
!
 
   

Because protocol-based T.38 fax relay directly involves Unified CM, make sure that your version of Unified CM supports T.38 fax relay within your gateway's call control protocol. To determine if your version of Cisco Unified CM supports T.38 fax relay with specific call control protocols, refer to the Cisco Unified Communications Manager release notes available at

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.html

In topologies that employ the Cisco 6608 or 6624 voice gateways in addition to other Cisco voice gateways that support T.38 fax relay, use the following Cisco IOS commands:

fax protocol t38 [nse [force]] fallback [cisco | none]
modem passthrough nse codec {g711ulaw|g711alaw}
 
   

These two commands enable Cisco IOS gateways to interoperate with the Cisco 6608 and 6624 gateways for Cisco fax relay and modem passthrough, as well as with other Cisco IOS gateways for T.38 fax relay and modem passthrough.

Gateways for Video Telephony

Video gateways terminate video calls into an IP telephony network or the PSTN. Deployments can consider separate gateways for voice calls and video calls, or they can have integrated gateways that route both voice and video calls.

Cisco offers voice gateway functionality in a variety of forms, such as standalone devices, modules that integrate into Cisco IOS Routers, or line cards that integrate into Cisco Catalyst Ethernet Switches. These gateways support multiple VoIP protocols (such as H.323, MGCP, SIP, and SCCP), multiple port interface types (such as FXS, FXO, E&M, T1/E1-CAS, T1/E1-PRI, ISDN BRI, and so on), and a myriad of advanced VoIP features. They also offer a rich set of management and troubleshooting interfaces. Cisco IOS routers also support H.320 gateway functionality and can be used as Integrated Voice and Video gateways.

Deployments that would like to leverage their existing voice infrastructure by adding video to it can do so by using the Cisco family of video-capable gateways from the Cisco Unified Videoconferencing 3500 Series portfolio.

The Unified Videoconferencing gateways, while excellent for video calls, do not support all of the features that Cisco Voice Gateways offer. The Unified Videoconferencing gateways have the following characteristics:

They support only H.323 and H.320.

They are standalone devices that cannot be integrated into Cisco IOS Routers or Cisco Catalyst Switches.

They support only T1/E1-PRI and ISDN BRI.

They support only G.711, G.722, G.722.1, G.723.1 and G.728; they do not support G.729 audio.

They support the H.245 Empty Capabilities Set (ECS).

They do not support many of the manageability and troubleshooting capabilities inherent in Cisco Voice Gateways.

As a result of these differences in the products, Cisco Unified Videoconferencing 3500 Series gateways are not recommended as replacements for Cisco Voice Gateways. IP Telephony customers who want to add video to their communications environment should deploy both types of gateways and use the Cisco Voice Gateways for all voice calls and use the Cisco Unified Videoconferencing 3500 Series gateways for video calls only. Customers might also have to procure separate circuits for voice and video from their PSTN service provider, depending on which model of Cisco IOS Gateway is deployed.

With separate voice and video gateways (see Figure 4-6), the route plans must also be separate for both inbound and outbound calls. For inbound calls, there is no way to have a single Direct Inward Dial (DID) extension for a user who wants to be able to receive both voice and video calls. Typically, each user will already have a DID for voice calls. When you introduce video into the scenario, users will have to be dialed some other way, such as via a second DID number or by dialing the main number of the video gateway and then entering the users video extension when prompted by the Interactive Voice Response (IVR). For outbound calls, there is no way to have a single PSTN access code for both voice and video calls. Typically, users will already have a well-known access code for voice (such as 9 in most US enterprises), but when you introduce video into the scenario, they will have to dial some other access code to place outbound video calls.

Figure 4-6 Unified CM System with Separate PSTN Lines for Voice and IP Video Telephony

Another consideration for deploying two types of gateways is the placement of those gateways. Typically, enterprises have many PSTN gateway resources consolidated at their central site(s), and each branch office has some local gateway resources as well. For instance, Cisco Catalyst 6500 gateways may be deployed at the central site with several T1/E1 circuits connected to them, while Cisco Integrated Services Routers (ISRs) may be deployed at each branch office with either analog or digital trunks to the local CO. When video is introduced into this scenario, the customer must also determine the number of PSTN circuits they will need for video and where the video gateways will be placed. For instance, will they deploy only a few IP/VC 3500 Series gateways at the central site, or will they also deploy them at each branch office?

Finally, consider how calls will be routed across the IP network to a remote gateway for the purpose of providing toll bypass, and how calls will be re-routed over the PSTN in the event that the IP network is unavailable or does not have enough bandwidth to complete the call. More specifically, do you want to invoke automated alternate routing (AAR) for video calls?

Routing Inbound Calls from the PSTN

Use one of the following methods to route inbound calls from the PSTN:

Assign at least two different directory numbers to each video-enabled device in the Unified CM cluster, with one line for audio and another line for video. With this method, the outside (PSTN) caller must dial the correct number to enable video.

For video calls, have outside callers dial the main number of the video gateway. Cisco Unified Videoconferencing gateways offer an integrated IVR that prompts the caller to enter the extension number of the party they are trying to reach. Unified CM will then recognize that it is a video call when ringing the destination device. This method relieves the caller from having to remember two different DID numbers for each called party, but it adds an extra step to dialing an inbound video call.


Note The outside video endpoints must support DTMF in order to enter the extension of the called party at the IVR prompt.


The following example illustrates the second method:

A user has a Cisco Unified IP Phone 7960 attached to a PC running Cisco Unified Video Advantage. The extension of the IP Phone is 51212, and the fully qualified DID number is 1-408-555-1212. To reach the user from the PSTN for a voice-only call, people simply dial the DID number. The CO sends calls to that DID number through T1-PRI circuit(s) connected to a Cisco Voice Gateway. When the call is received by the gateway, Unified CM knows that the gateway is capable of audio only, so it negotiates only a single audio channel for that call. Conversely, for people to reach the user from the PSTN for a video call, they must dial the main number of the video gateway and then enter the user's extension. For example, they might dial 1-408-555-1000. The CO would send calls to that number through the T1-PRI circuit(s) connected to a Cisco Unified Videoconferencing 3500 Series video gateway. When the call is received by the gateway, an IVR prompt asks the caller to enter the extension of the person they are trying to reach. When the caller enters the extension via DTMF tones, Unified CM knows that the gateway is capable of video, so it negotiates both audio and video channels for that call.

Gateway Digit Manipulation

The Cisco Unified Videoconferencing 3500 Series Gateways cannot manipulate digits for calls received from the PSTN. It takes the exact number of digits passed to it in the Q.931 Called Party Number field and sends them all to Unified CM. Therefore, Unified CM must manipulate the digits in order to match the directory number (DN) of the destination device. For instance, if the circuit from the CO switch to the gateway is configured to pass 10 digits but the extension of the called party is only five digits, Unified CM must strip off the leading five digits before attempting to find a matching DN. You can implement this digit manipulation in one of the following ways:

By configuring the Significant Digits field on the H.323 gateway device or on the H.225 gatekeeper-controlled trunk that carries the incoming calls from the IP/VC gateway

This method enables you to instruct Unified CM to pay attention to only the least-significant N digits of the called number. For example, setting the Significant Digits to 5 will cause Unified CM to ignore all but the last 5 digits of the called number. This is the easiest approach, but it affects all calls received from that gateway. Thus, if you have variable-length extension numbers, this is not the recommended approach.

By configuring a translation pattern and placing it in the calling search space of the H.323 gateway device or of the H.225 gatekeeper-controlled trunk that carries the incoming calls from the IP/VC gateway

This method enables Unified CM to match calls to the full number of digits received, to modify the called number, and then to continue performing digit analysis on the resulting modified number. This approach is slightly more complex than the preceding method, but it is more flexible and enables you to use a finer granularity for matching calls and for specifying how they will be modified.

Routing Outbound Calls to the PSTN

Use one of the following methods to route outbound calls to the PSTN:

Assign different access codes (that is, different route patterns) for voice and video calls. For example, when the user dials 9 followed by the PSTN telephone number they are trying to reach, it could match a route pattern that directs the call out a voice gateway. Similarly, the digit 8 could be used for the route pattern that directs calls out a video gateway.

Assign at least two different directory numbers on each video-enabled device in the Unified CM cluster, with one line for audio and another line for video. The two lines can then be given different calling search spaces. When users dial the access code (9, for example) on the first line, it could be directed out a voice gateway, while dialing the same access code on the second line could direct the call out a video gateway. This method alleviates the need for users to remember two different access codes but requires them to press the correct line on their phones when placing calls.

Gateway Service Prefixes

The Cisco Unified Videoconferencing Gateways use service prefixes to define the speed for outbound calls. When you configure a service prefix in the gateway, you must choose one of the following speeds:

Voice-only

128 kbps

256 kbps

384 kbps

768 kbps

Auto (dynamically determined; supports any call speed in the range of 128 kbps to 768 kbps)


Note Each of the above speeds represents a multiple of 64 kbps. For 56-kbps dialing, there is a check-box on the service prefix configuration page to restrict each channel to 56 kbps. Therefore, a 128-kbps service with restricted mode enabled would result in a 112-kbps service; a 384 kbps service with restricted mode enabled would result in a 336-kbps service; and so on.


Calls from an IP endpoint toward the PSTN must include the service prefix at the beginning of the called number in order for the gateway to decide which service to use for the call. Optionally, you can configure the default prefix to be used for calls that do not include a service prefix at the beginning of the number. This method can become quite complex because users will have to remember which prefix to dial for the speed of the call they wish to make, and you would have to configure multiple route patterns in Unified CM (one for each speed). Fortunately, the Auto speed enables you to minimize this effort. If the majority of your calls are made using 64 kbps per channel (for example, 128 kbps, 384 kbps, 512 kbps, 768 kbps, and so on), you could use the Auto service in that case. You would then need to create only one other service for the rare case in which someone makes a call using 56 kbps per channel (for example, 112 kbps, 336 kbps, and so on).

Cisco recommends that you always use a # character in your service prefixes because the gateway recognizes the # as an end-of-dialing character. By placing this character in the service prefix, you block people from attempting to use the gateway for toll fraud by dialing the main number of the gateway, reaching the IVR, and then dialing out to an off-net number. The # can either be at the beginning (recommended) or the end of the service prefix. For example, if your access code to reach the PSTN is 8 for video calls, Cisco recommends that you configure the service prefix as #8 or 8#. Or, if you have two service prefixes as described above, you might use #80 for the Auto 64-kbps service and #81 for the Auto 56-kbps service.

The ramification of using a service prefix is that Unified CM must prepend the service prefix to the called number when sending calls to the IP/VC gateway. Because forcing users to dial the # would not be very user-friendly, Cisco recommends that you configure Unified CM to prepend the # to the dialed number. For example, if the access code to dial a video call to the PSTN is 8, you could configure a route pattern as 8.@ in Unified CM, and in the route pattern configuration you would configure the called number translation rule to prepend #8 whenever that route pattern is dialed. Or, if you have two service prefixes as described above, you might use 80.@ for the Auto 64-kbps service (prefixing # to the called number) and 81.@ for the Auto 56-kbps service (prefixing # to the called number).

Automated Alternate Routing (AAR)

When the IP network does not have enough bandwidth available to process a call, Unified CM uses its call admission control mechanism to determine what to do with the call. As described in the chapter on IP Video Telephony, Unified CM performs one of the following actions with the call, depending on how you have configured it:

Fail the call, playing busy tone to the caller and displaying a Bandwidth Unavailable message on the caller's screen

Retry the video call as an audio-only call

Use automated alternate routing (AAR) to re-route the call over an alternative path, such as a PSTN gateway

The first two options are covered in the chapter on IP Video Telephony, and this section covers the AAR option.

To provide AAR for voice or video calls, you must configure the calling and called devices as members of an AAR group and configure an External Phone Number Mask for the called device. The External Phone Number Mask designates the fully qualified E.164 address for the user's extension, and the AAR group indicates what digits should be prepended to the External Phone Number Mask of the called device in order for the call to route successfully over the PSTN.

For example, assume that user A is in the San Jose AAR group and user B is in the San Francisco AAR group. User B's extension is 51212, and the External Phone Number Mask is 6505551212. The AAR groups are configured to prepend 91 for calls between the San Jose and San Francisco AAR groups. Thus, if user A dials 51212 and there is not enough bandwidth available to process the call over the IP WAN between those two sites, Unified CM will take user B's External Phone Number Mask of 6505551212, prepend 91 to it, and generate a new call to 916505551212 using the AAR calling search space for user A.

This same logic applies to video calls as well, with one additional step in the process. For video-capable devices, there is field called Retry Video Call as Audio. As described in the chapter on IP Video Telephony, if this option is enabled (checked), Unified CM does not perform AAR but retries the same call (that is, the call to 51212) as a voice-only call instead. If this option is disabled (unchecked), Unified CM performs AAR. By default, all video-capable devices in Unified CM have the Retry Video Call as Audio option enabled (checked). Therefore, to provide AAR for video calls, you must disable (uncheck) the Retry Video Call as Audio option. Additionally, if a call admission control policy based on Resource Reservation Protocol (RSVP) is being used between locations, the RSVP policy must be set to Mandatory for both the audio and video streams.

Furthermore, Unified CM looks at only the called device to determine whether the Retry Video Call as Audio option is enabled or disabled. So in the scenario above, user B's phone would have to have the Retry Video Call as Audio option disabled in order for the AAR process to take place.

Finally, devices can belong to only one AAR group. Because the AAR groups determine which digits to prepend, AAR groups also influence which gateway will be used for the rerouted call. Depending on your choice of configuration for outbound call routing to the PSTN, as discussed in the previous section, video calls that are rerouted by AAR might go out a voice gateway instead of a video gateway. Therefore, carefully construct the AAR groups and the AAR calling search spaces to ensure that the correct digits are prepended and that the correct calling search space is used for AAR calls.

While these considerations can make AAR quite complex to configure in a large enterprise environment, AAR is easier to implement when the endpoints are strictly of one type or the other (such as IP Phones for audio-only calls and systems such as the Tandberg T-1000 dedicated for video calls). When endpoints are capable of both audio and video calls (such as Cisco Unified Video Advantage or a Cisco IP Video Phone 7985G), the configuration of AAR can quickly become unwieldy. Therefore, Cisco recommends that large enterprise customers who have a mixture of voice and video endpoints give careful thought to the importance of AAR for each user, and use AAR only for select video devices such as dedicated videoconference rooms or executive video systems. Table 4-13 lists scenarios when it is appropriate to use AAR with various device types.

 

Table 4-13 When to Use AAR with a Particular Device Type 

Device Type
Device is used to call:
Enable AAR?
Comments

IP Phone

Other IP Phones and video-capable devices

Yes

Even when calling a video-capable device, the source device is capable of audio-only, thus AAR can be configured to route calls out a voice gateway.

IP Phone with Cisco Unified Video Advantage, or Cisco IP Video Phone 7985G

Other video-capable devices only

Yes

Because the device is used strictly for video calls, you can configure the AAR groups accordingly.

IP Phones and other video-capable devices

No

It will be difficult to configure the AAR groups to route audio-only calls differently than video calls.

Sony or Tandberg SCCP endpoint

Other video-capable devices only

Yes

Because the device is used strictly for video calls, you can configure the AAR groups accordingly.

IP Phones and other video-capable devices

No

It will be difficult to configure the AAR groups to route audio-only calls differently than video calls.

H.323 or SIP client

Other video-capable devices only

Yes

Because the device is used strictly for video calls, you can configure the AAR groups accordingly.

IP Phones and other video-capable devices

No

It will be difficult to configure the AAR groups to route audio-only calls differently than video calls


Least-Cost Routing

Least-cost routing (LCR) and tail-end hop-off (TEHO) are very popular in VoIP networks and can be used successfully for video calls as well. In general, both terms refer to a way of configuring the call routing rules so that calls to a long-distance number are routed over the IP network to the gateway closest to the destination, in an effort to reduce toll charges. (For Cisco Unified CM Release 4.1, LCR basically means the same thing as TEHO.) Unified CM supports this feature through its rich set of digit analysis and digit manipulation capabilities, including:

Partitions and calling search spaces

Translation patterns

Route patterns and route filters

Route lists and route groups

Configuring LCR for video calls is somewhat more complicated than for voice calls, for the following reasons:

Video calls require their own dedicated gateways, as discussed previously in this chapter

Video calls require much more bandwidth than voice calls

With respect to dedicated gateways, the logic behind why you might or might not decide to use LCR for video calls is very similar to that explained in the section on Automated Alternate Routing (AAR). Due to the need to have different types of gateways for voice and video, it can become quite complex to configure all the necessary partitions, calling search spaces, translation patterns, route patterns, route filters, route lists, and route groups needed for LCR to route voice calls out one gateway and video calls out another.

With respect to bandwidth requirements, the decision to use LCR depends on whether or not you have enough available bandwidth on your IP network to support LCR for video calls to/from a given location. If the current bandwidth is not sufficient, then you have to determine whether the benefits of video calls are worth the cost of either upgrading your IP network to make room for video calls or deploying local gateways and routing calls over the PSTN. For example, suppose you have a central site with a branch office connected to it via a 1.544-Mbps T1 Frame Relay circuit. The branch office has twenty video-enabled users in it. A 1.544-Mbps T1 circuit can handle at most about four 384-kbps video calls. Would it really make sense in this case to route video calls up to the central site in order to save on toll charges? Depending on the number of calls you want to support, you might have to upgrade your 1.544-Mbps T1 circuit to something faster. Is video an important enough application to justify the additional monthly charges for this upgrade? If not, it might make more sense to deploy an IP/VC video gateway at the branch office and not bother with LCR. However, placing local IP/VC gateways at each branch office is not inexpensive either, so ultimately you must decide how important video-to-PSTN calls are to your business. If video is not critical, perhaps it is not worth upgrading the bandwidth or buying video gateways but, instead, using the Retry Video Call as Audio feature to reroute video calls as voice-only calls if they exceed the available bandwidth. Once a call is downgraded to voice-only, local gateway resources and bandwidth to perform LCR become more affordable and easier to configure.

ISDN B-Channel Binding, Rollover, and Busy Out

There are two H.320 video channel bonding techniques, px64 and p*64 (ISO-13871). Cisco video equipment uses px64 for all calls, while other video equipment such as Polycom or Tandberg uses px64 for two-channel calls (128 kbps video) and ISO-13871 for anything higher than two channels (such as 192 kbps to 1 Mbps video). With Cisco IOS Release 12.4.20T, Cisco IOS H.320 gateways support the ISO-13871 bonding technique, which supports video calls at speeds up to 1 Mbps for video calls. With this functionality the Cisco IOS router can be used as an integrated gateways for both voice and video calls.

H.320 video uses multiple ISDN channels bound together to achieve the speeds needed to pass full-motion video. One of the problems with this bonding mechanism is that, when an inbound ISDN video call is received, the gateway does not know how many channels will be requested for that call until after it accepts the call and the source device indicates how many additional channels are required. If there are not enough B-Channels to satisfy the request, the call is disconnected. Therefore, careful traffic engineering is required to minimize the possibility that this situation will occur. Essentially, you want to ensure that there are always enough B-Channels available to handle the next call that might come in.

This B-Channel issue occurs in two cases:

Inbound calls from the PSTN to the IP network

Outbound calls from the IP network to the PSTN

Inbound Calls

For inbound calls, consider the following scenario:

A company has a Cisco 3526 IP/VC Gateway with an ISDN PRI circuit connecting it to a central office (CO) switch. The ISDN PRI circuit in this case offers 23 B-Channels. A video call is received from the PSTN at 384 kbps. This call takes six B-Channels, leaving 17 available. A second and third 384-kbps call are received on the line while the first one is still active. These each take an additional six channels, leaving five channels available. When the fourth 384-kbps call is received, the gateway will answer the call but, recognizing that it does not have enough B-Channels available (it only has five left but the call requires six), it will disconnect (by sending a Q.931 RELEASE COMPLETE with "16: Normal Call Clearing" as the reason). The caller attempting to make the fourth call will not know why the call failed and might redial the number repeatedly, trying to make the call work.

On Cisco Unified Videoconferencing gateways, you can minimize your chances of running into this issue by configuring the gateway to send a request to the CO to busy-out the remaining B-Channels (in this example, five channels) whenever the gateway reaches a certain threshold of utilization (configured as a percentage of total bandwidth).

In addition, you can have the CO provision multiple ISDN circuits in a trunk group. When the first circuit reaches the busy-out threshold, calls will roll over to the next PRI in the group. The Cisco 3540 IP/VC Gateway offers two ISDN PRI connections and supports bonding channels across both ports. For example, port 1 might have only five channels available while port 2 is sitting idle and, therefore, has 23 channels available. By taking the five channels from port 1 and one channel from port 2 and bonding them together, the fourth 384-kbps call can succeed. This leaves 22 channels available on controller 2, and at some point additional inbound calls would reach the busy-out threshold again. At that point the remaining channels on port 2 will be busied out, and all further inbound calls will be rejected with cause code "Network Congestion." Cisco Unified Videoconferencing gateways cannot bound channels across different gateways or across different Cisco 3540 gateway models in the same Cisco 3544 chassis, so two ports is the maximum that you can bond together. The CO switch can still roll calls over to a third or forth PRI in the trunk group (most COs support trunk groups of up to 6 circuits), but you cannot bond channels between PRI number one and PRI number three, for example, as you can between PRI number one and PRI number two.

The busy-out logic described above depends on the assumption that all calls take place at the same speed. Suppose, for example, that two 384-kbps calls are active on a port and a 128-kbps call came in. This call would take only two channels, using a total of 14 channels for the three calls (6+6+2 = 14) and leaving nine channels available on the circuit. However, if the busy-out threshold is set at 18 channels (assuming that all calls would take place at 384-kbps), only four channels are still available under this busy-out threshold. If another 384 kbps call comes in at this point, the call will fail because the remaining four channels are not enough to support the call. Also, because the busy-out threshold of 18 channels has not been reached yet (only 14 channels are used), the circuit is not busied out and calls will not roll over to the next circuit. This condition will persist until one of the existing calls is disconnected. To avoid such situations, it is important to try to standardize on a single call speed for all calls.

Outbound Calls

Outbound calls encounter the same potential situations as inbound calls, but the way in which the busy-out occurs is different. The Cisco 3500 Series IP/VC Gateways support messages called Resource Availability Indicator and Resource Availability Confirm (RAI/RAC). The RAI/RAC messages are defined under the H.225 RAS specification and are used by the gateways to tell the gatekeeper that they are full and to no longer route any more calls to them. When the gateway reaches the busy-out threshold, it sends an RAI message with a status of True to the gatekeeper. True means "Do not send me any more calls;" False means "I am available." The gateway sends an RAI=False as soon as it is no longer at its busy-out threshold. The busy-out threshold for outbound calls is separate from the busy-out threshold for inbound calls, and you can configure them differently so that inbound calls will roll over to the next available circuit but outbound calls will still be accepted, or vice versa. For example, you could configure the RAI threshold to 12 channels but the ISDN busy-out threshold to 18 channels. When two 384 kbps are active, outbound calls will roll over to the next available gateway, but a third 384-kbps inbound call could still be received. An equally efficient method of achieving outbound call busy-out failover is to use Unified CM's route group and route list construct, as described in the following section, instead of the RAI/RAC method.

Configuring the Gateways in Unified CM

You can configure a Unified Videoconferencing gateway in either of the following ways in Unified CM:

Configure it as an H.323 gateway, and Unified CM will route calls directly to the gateway.

Configure an H.225 gatekeeper-controlled trunk to the gatekeeper, and route calls to the gateway through the gatekeeper.

If you have only one gateway, it is probably easier to configure it directly in Unified CM instead of going through a trunk to get to it. If you have multiple gateways for load balancing and redundancy, you can either configure them all in Unified CM and place them into a route group(s) and route list, or configure an H.225 trunk to the gatekeeper and rely on RAI/RAC between the gateways and the gatekeeper to tell Unified CM which gateway it should send a given call to.

For inbound calls from the PSTN to Unified CM, the Cisco Unified Videoconferencing gateways can either register with a gatekeeper or be configured with the IP addresses of up to three Unified CM servers to which they should send all inbound call requests. This method is known as peer-to-peer mode. Either way, the goal is have all inbound calls received by the gateways sent to Unified CM so that Unified CM can decide how to route the calls. See Gatekeepers, for more details on how to configure the gatekeeper to route calls from the gateways to Unified CM.

Call Signaling Port Numbers

By default, the Cisco Unified Videoconferencing Gateways listen on TCP port 2720 instead of the well-known port 1720. However, also by default, Unified CM sends H.323 calls to port 1720. You can change the port that the gateway listens on or you can change the port that Unified CM sends to in the H.323 gateway device configuration in Unified CM. Either way, both sides have to match in order for outbound calls to the gateway to succeed.

In the inbound direction, when configured to operate in peer-to-peer mode, the Cisco Unified Videoconferencing Gateways will send the call to Unified CM on port 1720. When configured to register with a gatekeeper, Unified CM uses a randomly generated port number for all gatekeeper-controlled trunks. This method enables Unified CM to have multiple trunks to the same gatekeeper. This port number is included in the Registration Request (RRQ) from Unified CM to the gatekeeper, so the inbound H.225 setup message from the gateway to Unified CM will be sent to this port number. However, if the gateway is configured directly in Unified CM as an H.323 gateway device, Unified CM will ignore the fact that the call came in on the TCP port of the H.225 trunk and will instead match the source IP address to the H.323 gateway device configured in its database. If it does not find a matching device, Unified CM will treat the call as if it came in on the trunk.

In the outbound direction, if Unified CM uses a gatekeeper-controlled H.225 trunk to reach the gateway, the gatekeeper will tell Unified CM which TCP port to use to reach the gateway. If the gateway is configured in Unified CM as an H.323 gateway device (that is, peer-to-peer mode), then Unified CM must be configured to send calls either to port 2720 (default) or to 1720 (if the listening port on the gateway has been modified).

Call Signaling Timers

Due to the delay inherent in H.320 bonding, video calls can take longer to complete than voice calls. Several timers in Unified CM are tuned, by default, to make voice calls process as fast as possible, and they can cause video calls to fail. Therefore, you must modify the following timers from their default values in order to support H.320 gateway calls:

H.245TCSTimeout

Media Exchange Interface Capability Timer

Media Exchange Timer

Cisco recommends that you increase each of these timers to 25 by modifying them under the Service Parameters in Unified CM Administration. Note that these are cluster-wide service parameters, so they will affect calls to all types of H.323 devices, including voice calls to existing H.323 Cisco Voice Gateways.

Bearer Capabilities of Voice Gateways

H.323 calls use the H.225/Q.931 Bearer Capabilities Information Element (bearer-caps) to indicate what type of call is being made. A voice-only call has its bearer-caps set to "speech" or "3.1 KHz Audio" while a video call has its bearer-caps set to "Unrestricted Digital Information." Some devices do not support Unrestricted Digital Information bearer-caps. Calls to these devices might fail if Unified CM attempts the call as a H.323 video call.

Unified CM decides which bearer-caps to set, based on the following factors:

Whether the calling and/or called devices are video-capable

Whether the region in Unified CM is configured to allow video for calls between those devices

Unified CM supports retrying the video call as audio, and this feature can be enabled through configuration. When Unified CM makes a video call with bearer-caps set to "Unrestricted Digital" and the call fails, Unified CM then retries the same call as an audio call with the bearer-caps set to "speech."

When using H.323, Cisco IOS gateways can service calls as voice or video, based on the bearer capabilities it receives in the call setup. When using SIP, the gateway translates the ISDN capabilities into SDP for call negotiations.

If the Cisco voice gateway uses MGCP to communicate with Unified CM, the problem will not occur because Unified CM does not support video on its MGCP protocol stack and because, in MGCP mode, Unified CM has complete control over the D-Channel signaling to the PSTN.