Interworking Signaling Enhancements for H.323 VoIP
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Table Of Contents
Interworking Signaling Enhancements for
H.323 VoIPRelated Features and Technologies
Supported Standards, MIBs, and RFCs
Configuring Progress Indicator in H.323 VoIP Dial Peers
Configuring Progress Indicator in H.323 POTS Dial Peers
Verifying Progress Indicator Configuration
Configuring ISDN T306 and T310 Timers
Verifying T306 Timer Configuration
Progress Indicator Configuration Example
T306/T310 Timer Configuration Example
Interworking Signaling Enhancements for
H.323 VoIP
Feature History
12.1(3)XI
This feature was introduced.
12.1(5)XM2
Support was added for the Cisco AS5350 and Cisco AS5400 universal gateways.
This feature module describes enhancements to H.323 and Session Initiation Protocol (SIP) signaling when interworking with ISDN, T1 channel associated signaling (CAS), or E1 R2 Public Switched Telephone Network (PSTN) services. These enhancements improve the call signaling capabilities between the Cisco VoIP gateway and the telco switch to ensure, for example, that the voice path is completed (cut-through) at the appropriate point in call setup and that early alerting (ringing) does not occur.
This document includes the following sections:
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Supported Standards, MIBs, and RFCs
Feature Overview
The Interworking Signaling Enhancements for H.323 and SIP VoIP feature enables VoIP networks to properly signal the setup and tear-down of calls, including generating in-band tones and announcements when needed at the originating or terminating switch. When a tone (for example, ringback, busy, reorder) or announcement (for example, "The number you have dialed is no longer in service") is played at the destination switch, the backward voice path from the called party to the calling party is cut-through early, so that the calling party can hear the tone or announcement. To prevent fraudulent calls, the voice path is cut-through in both directions only after the Connect message is received from the destination. The call progress indicator, which signals the availability of in-band communication, is carried end-to-end as required when interworking with ISDN and CAS protocols.
These enhancements prevent unexpected behavior such as early alerting (when an Alert message is returned immediately after a Call Proceeding message is sent), to ensure that the calling party does not hear conflicting call progress information such as a ringback tone followed by a busy tone, and does not miss hearing a tone or announcement when one should play. In addition, support for network-side ISDN and reducing the risk of speech clipping is addressed.
This feature set provides:
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In-Band Tones and Announcements
In-band progress tones and announcements are required for PSTN services and for ISDN speech and 3.1 kHz audio services, per Bellcore and ANSI specifications. To guarantee that in-band tones and announcements are generated when required and at the appropriate switch, this feature set ensures that the progress indicator (PI) is carried end-to-end in call signaling messages between the called party and the calling party. You are also allowed to configure the progress indicator in outbound dial peers at the H.323 VoIP gateway, if necessary.
The progress indicator is an information element (IE) that signals when in-band tones and announcements are available. The progress indicator controls whether the local switch generates the appropriate tone or announcement or whether the remote switch is responsible. For example, if the destination switch generates the ringback tone, it sends a progress indicator of 1 or 8 in the Alerting message. If the origination switch receives an Alerting message without a progress indicator, it generates the ringback tone.
The specific progress indicator that a switch sends in call messages, if any, depends on the model of the switch. To ensure that in-band communication is generated appropriately, it may be necessary in some instances to override the default behavior of the switch by manually configuring the progress indicator at the Cisco H.323 gateway.
The progress indicator is configurable in Setup messages from the outbound VoIP dial peer, typically at the originating gateway, and in Progress and Connect messages from the outbound POTS dial peer, typically at the terminating gateway. The progress indicator is configured by using the progress_ind dial-peer configuration command.
Table 1 shows the progress indicator values that you can configure through Cisco IOS on the H.323 gateway for the different types of messages.
When interworking between ISDN and non-ISDN networks:
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Note
End-to-End Alerting
Early alerting is prevented in these ways:
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Cut-through of Voice Path
When tones and announcements are generated at the destination switch, the backward voice path from the called party to the calling party is cut-through before the tones and announcements are played. This allows announcements such as, "The number you have called has been changed" or tones for error conditions, such as network congestion, to be forwarded to the calling party. To prevent fraudulent calls, the originating gateway does not perform full cut-through until it receives a Connect message from the destination switch. Cut-through is performed as follows:
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ISDN Cause Codes
The cause code is an information element (IE) that indicates why an ISDN call failed or was otherwise disconnected. When the originating gateway receives a Release Complete message, it generates the appropriate tone based on the cause code in the message.
Table 2 lists the default cause codes that the VoIP gateway sends to the switch when a call fails at the gateway, and the corresponding tones that it generates.
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For a complete list of ISDN cause codes that are generated by the switch, see Appendix B in the Cisco IOS Debug Command Reference, Cisco IOS Release 12.1.
Although the VoIP gateway generates the cause codes listed in Table 2 by default, there are commands introduced in previous Cisco IOS releases that can override these defaults, allowing the gateway to send different cause codes to the switch.
The following commands override the default cause codes:
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When you implement these commands, the configured cause codes are sent to the switch; otherwise, the default cause codes of the voice application are sent. For a complete description of these commands, see the Cisco IOS Dial Services Command Reference, Cisco IOS Release 12.1.
ISDN T306 Disconnect Timer and T310 Timer
A new disconnect timer, T306, has been added to allow in-band announcements and tones to be played before a call is disconnected. It is designed for routers that are configured as an ISDN network-side switch. The T306 timer starts when the gateway receives a disconnect message with a progress indicator of 8. The voice path is cut-through in the backward direction, and the announcement or error tone is played until the timer expires. When the timer expires, the voice application disconnects the call. You can configure this timer by using the isdn t306 command.
The T310 timer sets a limit for a call in the Call Proceeding state. The timer starts when the router receives a Call Proceeding message and stops when the call moves to another phase, typically Alerting, Connect, or Progress. If the timer expires while the call is in the Call Proceeding state, the router releases the call. You can configure this timer by using the isdn t310 command.
H.245 Initiation
To avoid speech clipping, H.245 capabilities are now initiated at the originating gateway at the earliest possible moment, when the originating gateway receives a Call Proceeding message from the terminating gateway. Previously, Call Proceeding messages were not passed end-to-end across the VoIP network; H.245 was initiated only after the originating gateway received an Alert message.
Overlap Dialing
To enhance overlap dialing, the Call Proceeding message is now passed transparently from the terminating switch to the originating switch, when the originating switch does not include the Sending Complete information element in the Setup message. The Call Proceeding message notifies the originating switch that the terminating switch has collected all digits that are required to route the call. If the originating switch sends a Sending Complete IE, the originating gateway responds with a Call Proceeding message, and the session application drops the Call Proceeding message sent by the terminating switch.
SIP 183 Session Progress Message
SIP 183 Session Progress messages are supported, facilitating better call treatment for SIP VoIP calls when interworking with PSTN networks. The introduction of the 183 Session Progress message allows a called user agent to suppress local alerting from the calling user agent, and to play a tone or announcement during a preliminary call session, before the full SIP session is set up. This enables the calling party to be notified of the status of the call without being charged for the preliminary portion of the call. A new Session header in the 183 Session Progress message controls whether or not the called user agent plays a tone or announcement for the calling party. The 183 Session Progress message is supported by default and does not require any special configuration.
Table 3 lists ISDN and CAS messages that are sent by the switch, and the corresponding SIP messages that the gateway generates in response.
Table 4 lists the SIP messages that are generated by the gateway, and the corresponding ISDN and CAS messages that the switch produces in response.
Benefits
This feature set ensures that the call signaling for VoIP services is handled properly when interworking with CAS and ISDN networks, resulting in:
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Restrictions
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Related Features and Technologies
These features are dependent on the interoperability of Service Provider Features for VoIP.
Related Documents
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Supported Platforms
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Supported Standards, MIBs, and RFCs
Standards
No new or modified standards are supported by this feature.
MIBs
No new or modified MIBs are supported by this feature.
To obtain lists of MIBs supported by platform and Cisco IOS release and to download MIB modules, go to the Cisco MIB web site on Cisco Connection Online (CCO) at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
RFCs
No new or modified RFCs are supported by this feature.
Prerequisites
The Cisco AS5350 and Cisco AS5400 do not support the Mica Modem Card, Microcom Modem Card, or VoIP Feature Card. Voice and modem functions are provided by the Universal Port Dial Feature card running SPE firmware. See the Cisco AS5350 Universal Gateway Card Installation Guide and the Cisco AS5400 Universal Gateway Card Installation Guide for more information. All references to the Cisco AS5300 in this document apply to the Cisco AS5350 and Cisco AS5400 platforms with the following exceptions:
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Other Prerequisites
To use these features, you must first:
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Cisco IOS IP and IP Routing Configuration Guide, Cisco IOS Release 12.1.Configuration Tasks
See the following sections for configuring these optional signaling interworking features:
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Configuring Progress Indicator in H.323 VoIP Dial Peers
Note
To include a specific progress indicator in Setup messages from the outbound VoIP dial peer on an H.323 gateway, perform the following tasks in dial-peer configuration mode:
Configuring Progress Indicator in H.323 POTS Dial Peers
Note
To include a specific progress indicator in Progress or Connect messages from the outbound POTS dial peer on an H.323 gateway, perform the following tasks in dial-peer configuration mode:
Verifying Progress Indicator Configuration
Perform the following steps to verify that the progress indicator is configured and operating correctly.
Step 1
Step 2
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Configuring ISDN T306 and T310 Timers
To configure the T306 and T310 timers, perform the following tasks in interface configuration mode:
Verifying T306 Timer Configuration
Perform the following steps to verify that the T306 timer is configured and operating correctly.
Step 1
Step 2
Step 3
Troubleshooting Tips
The following table lists some potential configuration issues and their resolutions.
Calling party does not hear ringback tone after alerting
Enable debug isdn q931 command to display the ISDN messages. Verify the progress indicator value in the Setup message from the originating switch. If the Setup message does not contain a PI value, use the progress_ind dial-peer command to set the PI to 3.
Gateway not responding to Connect after receiving Progress
Enable debug isdn q931 command to display the ISDN messages. Verify that a progress indicator is included in the Progress message and that the PI meets ITU-T standards.
Configuration Examples
This section provides the following configuration examples:
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Progress Indicator Configuration Example
!dial-peer voice 3 potsdestination-pattern 55275session protocol ciscoprogress_ind progress enable 1progress_ind connect enable 1port 1:0!..T306/T310 Timer Configuration Example
!interface Serial0:23no ip addressno ip directed-broadcastencapsulation pppdialer rotary-group 0isdn switch-type primary-5essisdn incoming-voice modemisdn t306 60000isdn t310 40000!..Command Reference
This section documents new commands. All other commands used with this feature are documented in the Cisco IOS Release 12.1 command reference publications.
isdn negotiate-bchan
To enable the router to accept a B-channel that is different from the B-channel requested in the outgoing call setup message, use the isdn negotiate-bchan interface configuration command. To restore the default condition, use the no form of this command.
isdn negotiate-bchan
no isdn negotiate-bchan
Syntax Description
This command has no keywords or arguments.
Defaults
The router will not accept a different B-channel.
Command Modes
Interface configuration
Command History
11.3(1)T
This command was introduced.
12.1(5)XM2
The command was introduced for the Cisco AS5350 and CiscoAS5400.
Usage Guidelines
The isdn negotiate-bchan command is supported only for NET5 and N12 switches.
Examples
The following example enables an alternate B-channel for serial interface 0:23:
interface serial0:23isdn negotiate-bchanRelated Commands
isdn t306
To set a timer for disconnect messages received by the router, use the isdn t306 interface configuration command. To restore the default value, use the default or no form of this command.
isdn t306 msecs
default isdn t306
no isdn t306
Syntax Description
msecs
Number of milliseconds that the router waits before disconnecting a call after it receives a disconnect message with a progress indicator of 8. Values are 1 through 400,000 ms.
Defaults
The default depends on the switch, usually from 5,000 to 30,000 ms.
Command Modes
Interface configuration
Command History
12.1(3)XI
This command was introduced.
12.1(5)XM2
The command was introduced for the Cisco AS5350 and CiscoAS5400.
Usage Guidelines
The T306 timer is designed for routers that are configured as an ISDN network-side switch. When the router receives a disconnect message with a progress indicator of 8, it disconnects the call after waiting for the specified number of ms while the in-band announcement or error tone is playing. Be sure to set the timer long enough for the announcement to be heard or the tone to be recognized. The isdn t306 command is used only for disconnect messages with a progress indicator of 8; otherwise, the T305 timer is used. The disable and no forms of this command have the same result: the timer waits for the default number of ms before disconnecting the call.
Examples
The following example sets the T306 timer to 60000 ms for serial interface 0:23:
interface serial 0:23isdn t306 60000Related Commands
isdn t310
To set a timer for the Call Proceeding state, use the isdn t310 interface configuration command. To restore the default value, use the no form of this command.
isdn t310 msecs
no isdn t310
Syntax Description
msecs
Number of milliseconds that the router waits before disconnecting a call after receiving a Call Proceeding message. Values are 1 through 400,000 ms.
Defaults
The default depends on the switch, usually from 5,000 to 30,000 ms.
Command Modes
Interface configuration
Command History
12.1(3)XI
This command was introduced.
12.1(5)XM2
The command was introduced for the Cisco AS5350 and CiscoAS5400.
Usage Guidelines
The T310 timer starts when the router receives a Call Proceeding message; it stops when the call exits the Call Proceeding state, typically when the call moves to Alerting, Connect, or Progress. If the timer expires while the call is in the Call Proceeding state, the router releases the call. Set the timer to match the specific characteristics of your network.
Examples
The following example sets the T310 timer to 40,000 ms for serial interface 0:23:
interface serial 0:23isdn t310 40000Related Commands
progress_ind
To set a specific progress indicator in call Setup, Progress, or Connect messages from an H.323 VoIP gateway, use the progress_ind dial-peer configuration command. To restore the default condition, use the no or disable forms of this command.
progress_ind {setup | connect | progress} {enable pi-number | disable}
no progress_ind {setup | connect | progress}
Note
Syntax Description
Defaults
The default progress indicator from the switch is not intercepted or modified.
Command Modes
Dial-peer configuration
Command History
12.1(3)XI
This command was introduced.
12.1(5)XM2
The command was introduced for the Cisco AS5350 and CiscoAS5400.
Usage Guidelines
The progress_ind command overrides the default progress indicator that is sent by the switch. This enables you to set the progress indicator at the H.323 gateway, if necessary, to ensure the proper end-to-end signaling for VoIP calls. This command sets the progress indicator only in messages from outbound dial peers that have a set destination pattern, configured by using the destination-pattern command. If a message contains multiple progress indicators, the progress_ind command overrides only the first progress indicator in the message.
The disable and no forms of the progress_ind command have the same result: the call messages are not intercepted by the session application, and the default progress indicator, if any, is forwarded unmodified.
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Examples
The following example sets the progress indicator to 1 in Progress and Connect messages from the number 3 POTS dial peer:
dial-peer voice 3 potsdestination-pattern 55275progress_ind progress enable 1progress_ind connect enable 1Related Commands
Debug Commands
This section documents a new debug command. All other commands used with this feature are documented in the Cisco IOS Release 12.1 command reference publications.
debug vtsp tone
To display debug messages showing the types of tones generated by the VoIP gateway, use the debug vtsp tone command. To disable the debug messages, use the no form of this command.
debug vtsp tone
no debug vtsp tone
Syntax Description
This command has no keywords or arguments.
Defaults
Tone generation messages are not enabled.
Command History
12.1(3)XI
This command was introduced.
12.1(5)XM2
The command was introduced for the Cisco AS5350 and CiscoAS5400.
Examples
The following example shows that a ringback tone was generated by the VoIP gateway:
Router# debug vtsp tone*Jan 1 16:33:52.395:act_alert:Tone Ring Back generated in direction Network*Jan 1 16:33:52.399:ISDN Se0:23:TX -> ALERTING pd = 8 callref = 0x9816Related Commands
Glossary
CAS—channel associated signaling. Call signaling that enables the access server to send or receive analog calls or calls on digital trunks using robbed-bit signaling.
cause code—Defined by ITU Recommendation Q.850; indicates the reason for ISDN call failure or completion.
cut-through—Completion of the bearer transmission path between the calling party and the called party.
dial peer—An addressable call endpoint. In Voice over IP (VoIP), there are two types of dial peers: POTS and VoIP.
gateway—A gateway allows H.323 terminals to communicate with non-H.323 terminals by converting protocols. A gateway is the point at which a circuit-switched call is encoded and repackaged into IP packets.
H.323—An International Telecommunication Union (ITU-T) standard that describes packet-based video, audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the conferencing system and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual protocol.
interworking—The mapping of call signaling messages between two different protocol suites.
ISDN—Integrated Services Digital Network. Communication protocol offered by telephone companies that permits telephone networks to carry data, voice, and other source traffic.
POTS—Plain old telephone service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the PSTN.
POTS dial peer—Dial peer connected by a traditional telephony network. POTS peers point to a particular voice port on a voice network device.
progress indicator—An information element (IE) in ISDN messages that indicates when in-band communication is used.
PSTN—Public Switched Telephone Network. General term referring to the variety of telephone networks and services in place worldwide. Sometimes called plain old telephone service (POTS) .
R2—Channelized E1 signaling used in Europe, Asia, and South America. It is equivalent to channelized T1 signaling in North America.
SIP—Session Initiation Protocol. This is a protocol developed by the IETF MMUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999. SIP equips platforms to signal the setup of voice and multimedia calls over IP networks.
VoIP—Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with POTS-like functionality, reliability, and voice quality. VoIP is a blanket term, that generally refers to the Cisco standards-based (for example H.323) approach to IP voice traffic.
VoIP dial peer—Dial peer connected by a packet network; in the case of Voice over IP, this is an IP network. VoIP peers point to specific VoIP devices.
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