Table Of Contents
Interworking Signaling Enhancements for
H.323 and SIP 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
Configuring Disconnect With PI
Configuring Maximum Interworking
Configuring Two-Way Voice Path for RTP
Verifying Disconnect with PI, Maximum Interworking, and Two-Way RTP
Progress Indicator Configuration Example
T306/T310 Timer Configuration Example
Disconnect PI Configuration Example
Maximum Interworking and Two-Way RTP Configuration Example
Interworking Signaling Enhancements for
H.323 and SIP VoIP
Feature History
This feature module describes enhancements to H.323 and Session Initiation Protocol (SIP) signaling when interworking with ISDN, T1 channel associated signaling (CAS), and E1 R2 services from the Public Switched Telephone Network (PSTN). 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 established (cut-through) at the appropriate point during 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 can also 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 terminating switch generates the ringback tone, it sends a progress indicator of 1 or 8 in the Alerting message. If the originating 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 Alert, 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 (PI) values that you can configure through Cisco IOS software on the H.323 gateway for the different types of messages.
When interworking between ISDN and non-ISDN networks:
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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.
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, if 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 dialed 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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When the Cisco AS5300 and the SC2200 receive a Resume message, the timers are stopped. If either of the timers expire, the call is released with a cause code of normal clearing.
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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These features run on all platforms that support Cisco IOS Release 12.1(5)T and VoIP features.
Determining Platform Support Through Cisco Feature Navigator
Cisco IOS software is packaged in feature sets that support specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.
Cisco Feature Navigator is a web-based tool that enables you to determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.
To access Cisco Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions at http://www.cisco.com/register.
Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
Availability of Cisco IOS Software Images
Platform support for particular Cisco IOS software releases is dependent on the availability of the software images for those platforms. Software images for some platforms may be deferred, delayed, or changed without prior notice. For updated information about platform support and availability of software images for each Cisco IOS software release, refer to the online release notes or, if supported, Cisco Feature Navigator.
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 supported MIBs 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
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 beginning in global configuration mode:
Configuring Progress Indicator in H.323 POTS Dial Peers
Note
To include a specific progress indicator in Alert, Progress, or Connect messages from the outbound POTS dial peer on an H.323 gateway, perform the following tasks beginning in global configuration mode:
Verifying Progress Indicator Configuration
Perform the following steps in privileged EXEC mode 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 in privileged EXEC mode to verify that the T306 timer is configured and operating correctly.
Step 1
Step 2
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Configuring Disconnect With PI
To enable the H.323 gateway to treat a Disconnect message with a progress indicator (PI) the same as a standard Disconnect, perform the following tasks beginning in global configuration mode:
Configuring Maximum Interworking
To enable maximum interworking on the H.323 gateway, perform the following task in global configuration mode:
Configuring Two-Way Voice Path for RTP
To enable a two-way voice path when the Real-Time Transport Protocol (RTP) channel is opened, perform the following task in global configuration mode:
Router(config)# voice rtp send-recv
Establishes a two-way voice path when the RTP channel is opened.
Verifying Disconnect with PI, Maximum Interworking, and Two-Way RTP
To verify that the configuration is accurate, display the running configuration file by using the show running-config command.
Troubleshooting Tips
The following table lists some potential configuration issues and their resolutions.
Calling party does not hear ringback tone after alerting
Enable the debug isdn q931 command to display the ISDN messages and verify the progress indicator values. Do one of the following depending on where you want to generate the ringback tone:
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If the Setup message from the originating switch does not contain a PI value, use the progress_ind dial-peer command to set the PI to 3.
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If the Alert message from the terminating switch does not contain a PI value, use the progress_ind dial-peer command to set the PI to 8 at the terminating gateway.
Gateway not responding to Connect after receiving Progress
Enable the 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.
Interoperability Issues with Legacy H.323 Gateways
For H.323 gateways that are running Cisco IOS Release 12.1(2)T or earlier:
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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..Disconnect PI Configuration Example
!voice-port 0:Ddisc_pi_off!voice-port 1:D!voice-port 3:D..Maximum Interworking and Two-Way RTP Configuration Example
!async-bootp dns-server 172.22.53.210isdn switch-type primary-5essisdn voice-call-failure 0isdn alert-end-to-endisdn gateway-max-interworking!cns event-service servervoice call send-alertvoice rtp send-recv..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.
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disc_pi_off
To enable an H.323 gateway to disconnect a call when it receives a Disconnect message with a progress indicator (PI) value, use the disc_pi_off voice-port configuration command. To restore the default value, use the no form of this command.
disc_pi_off
no disc_pi_off
Syntax Description
This command has no keywords or arguments.
Defaults
The gateway does not disconnect a call when it receives a Disconnect message with a PI value.
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The disc_pi_off voice-port command is only valid if the Disconnect with PI is received on the inbound call leg. For example, if this command is enabled on the voice port of the originating gateway, and a Disconnect with PI is received from the terminating switch, the Disconnect with PI is converted to a Disconnect. But if this command is enabled on the voice port of the terminating gateway, and a Disconnect with PI is received from the terminating switch, the Disconnect message is not converted to a standard Disconnect, because the Disconnect message is received on the outbound call leg.
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Examples
The following example handles a Disconnect message with a PI value the same as a standard Disconnect message for voice port 0:23:
voice-port 0:Ddisc_pi_offRelated Commands
isdn gateway-max-interworking
To prevent the H.323 gateway from checking for ISDN protocol compatibility and dropping information elements (IEs) in call messages, use the isdn gateway-max-interworking global configuration command. To restore the default condition, use the no form of this command.
isdn gateway-max-interworking
no isdn gateway-max-interworking
Syntax Description
This command has no keywords or arguments.
Defaults
The gateway checks for protocol compatibility
Command Modes
Global configuration
Command History
Usage Guidelines
If the isdn gateway-max-interworking command is enabled on the originating H.323 gateway, the information elements (IEs) in call messages to the terminating gateway are not checked for end-to-end protocol compatibility. If this command is enabled on the terminating gateway, IEs are not checked in the reverse direction. If this command is not enabled, and the ISDN protocols are not compatible on the originating and terminating gateways, the gateway drops all IEs, including the progress indicator. The gateway then inserts a progress indicator of 1 into all Progress messages.
Examples
The following example enables maximum interworking:
isdn gateway-max-interworkingisdn 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 to 400000 ms.
Defaults
The default depends on the switch, usually from 5000 to 30000
Command Modes
Interface configuration
Command History
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 to 400000 ms.
Defaults
The default depends on the switch, usually from 5000 to 30000
Command Modes
Interface configuration
Command History
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 (PI) 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 | alert} {enable pi-number | disable}
no progress_ind {setup | connect | progress | alert}
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Syntax Description
Defaults
The default progress indicator from the switch is not intercepted or modified.
Command Modes
Dial-peer configuration
Command History
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
voice rtp send-recv
To establish a two-way voice path when the Real-Time Transport Protocol (RTP) channel is opened, use the voice rtp send-recv global configuration command. To restore the default condition, use the no form of this command.
voice rtp send-recv
no voice rtp send-recv
Syntax Description
This command has no keywords or arguments.
Defaults
The voice path is cut-through in only the backward direction when the RTP channel is opened.
Command Modes
Global configuration
Command History
Usage Guidelines
The voice rtp send-recv command should be enabled only when the voice path must be cut-through (established) in both the backward and forward directions before a Connect message is received from the destination switch. The voice rtp send-recv command affects all VoIP calls when it is enabled.
Examples
The following example enables the voice path to cut-through in both directions when the RTP channel is opened:
voice rtp send-recvDebug Commands
This section documents the new debug vtsp tone 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
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
Guest
