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Table Of Contents
Related Features and Technologies
Supported Standards, MIBs, and RFCs
Configuring the SIP Media Inactivity Timer
Verifying the SIP Media Inactivity Timer
Configuring the SIP Media Inactivity Timer Feature Example
SIP Media Inactivity Timer
Document Update Alert
This document was originally produced for Cisco IOS Release 12.2(11)T. This feature has been updated in subsequent releases, and more recent documentation is available.
If you are using Cisco IOS Release 12.2(11)T or higher, refer to the following section in the Configuring Additional SIP Features chapter of the Cisco IOS SIP Configuration Guide, Cisco IOS Voice Configuration Library, Release 12.3:
Feature History
This document describes the Session Initiation Protocol (SIP) Media Inactivity Timer feature. It includes the following sections:
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Supported Standards, MIBs, and RFCs
Feature Overview
The SIP Media Inactivity Timer feature enables Cisco gateways to monitor and disconnect Voice-over-IP (VoIP) calls if no Real-Time Control Protocol (RTCP) packets are received within a configurable time period.
When RTCP reports are not received by a Cisco gateway, the SIP Media Inactivity Timer feature releases the hung session and its network resources in an orderly manner. These network resources include the gateway digital signal processor (DSP) and time-division multiplexing (TDM) channel resources that are utilized by the hung sessions. Because call signaling is sent to tear down the call, any stateful SIP proxies involved in the call are also notified to clear the state that they have associated with the hung session. The call is also cleared back through the TDM port so that any attached TDM switching equipment also clears its resources.
Benefits
Provides a mechanism for detecting and freeing hung network resources when no RTCP packets are received by the gateway.
Related Features and Technologies
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Related Documents
The following documents contain information related to the Cisco SIP functionality:
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Supported Platforms
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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
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To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL:
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
RFCs
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Configuration Tasks
See the following sections for configuration tasks for this feature. Each task in the list is identified as either required or optional.
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Configuring the SIP Media Inactivity Timer
The SIP Media Inactivity Timer feature requires the configuration of the ip rtcp report interval command and the timer receive-rtcp command to enable detection of RTCP packets by the gateway. When these commands are configured, the gateway uses RTCP report detection, rather than Real-Time Protocol (RTP) packet detection, to determine whether calls on the gateway are still active or should be disconnected. This method is more reliable because there are periods during voice calls when one or both parties are not sending RTP packets.
One common example of a voice session in which no RTP is sent is when a caller dials into a conference call and mutes his endpoint. If voice activity detection (VAD, also known as silence suppression) is enabled, no RTP packets are sent while the endpoint is muted. However, the muted endpoint continues to send RTCP reports at the interval specified by the ip rtcp report interval command.
The timer receive-rtcp value argument (or Mfactor) is multiplied with the interval that is set using the ip rtcp report interval command. If no RTCP packets are received in the resulting time period, the call is disconnected. The gateway signals the disconnect to the SIP network and the TDM network so that upstream and downstream devices can clear their resources. The gateway sends a SIP BYE to disconnect the call and sends a Q.931 DISCONNECT back to the TDM network to clear the call upon the expiration of the timer. The Q.931 DISCONNECT is sent with a Cause code value of 3 (no route). There is no Q.931 Progress Indicator (PI) value included in the DISCONNECT.
To configure the SIP Media Inactivity Timer feature, enter the following commands beginning in global configuration mode:
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Verifying the SIP Media Inactivity Timer
To verify that the SIP Media Inactivity Timer feature is enabled, follow these steps:
Step 1
Step 2
Troubleshooting Tips
To troubleshoot the SIP Media Inactivity Timer feature, perform the following tasks:
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Router# debug ccsip events00:04:29: sipSPICreateAndStartRtpTimer: Valid RTP/RTCP session found and CLI enabled to create and start the inactivity timer00:04:29: sipSPICreateAndStartRtpTimer:Media Inactivity timer created for call.Mfactor(from CLI): 5 RTCP bandwidth: 500RTCP Interval(in ms): 5000Normalized RTCP interval (in ms):25000The following example trace shows a timer expiring:
Router# debug ccsip events02:41:03: %LINEPROTO-5-UPDOWN: Line protocol on Interface Ethernet0, changed state to down*Jan 1 02:41:34.107: sipSPIRtpDiscTimerExpired:RTP/RTCP receive timer expired. Disconnect the call.*Jan 1 02:41:34.107: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT*Jan 1 02:41:34.107: CCSIP-SPI-CONTROL: act_active_disconnect
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Configuration Examples
This section provides the following configuration example:
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Configuring the SIP Media Inactivity Timer Feature Example
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Router# show running-config!version 12.2no parser cacheservice timestamps debug datetime msecservice timestamps log uptimeno service password-encryption!hostname madisonboot system flashno logging bufferedaaa new-model!aaa authentication login h323 group radiusaaa authorization exec h323 group radiusaaa accounting connection passsword stop-only group radiusaaa accounting connection h323 start-stop group radiusaaa session-id common!resource-pool disableclock timezone EST -5!ip subnet-zeroip tcp path-mtu-discoveryip name-server 172.18.192.48ip dhcp smart-relay!isdn switch-type primary-ni!voice service voiph323!voice class codec 1codec preference 1 g723ar53codec preference 2 g723r53codec preference 3 g729br8codec preference 4 gsmfrcodec preference 5 g726r24codec preference 6 g726r32voice class codec 2codec preference 1 g729br8codec preference 2 g729r8codec preference 3 g723ar53codec preference 4 g723ar63codec preference 5 g723r53codec preference 6 g723r63codec preference 7 gsmfrcodec preference 8 gsmefr!voice class codec 3codec preference 1 g726r24codec preference 2 gsmefrcodec preference 3 g726r16!fax interface-type modemmta receive maximum-recipients 0controller T1 0framing esfclock source line secondary 1linecode amipri-group timeslots 1-24description summa_pbx!controller T1 1framing esflinecode amipri-group timeslots 1-24description summa_pbx!controller T1 2framing sflinecode ami!controller T1 3framing esfclock source line primarylinecode b8zsds0-group 0 timeslots 1-24 type e&m-fgb dtmf dniscas-custom 0!gw-accounting h323 vsagw-accounting voipinterface Ethernet0ip address 172.18.193.99 255.255.255.0no ip route-cacheno ip mroute-cacheip rsvp bandwidth 7500 7500!interface Serial0:23no ip addressisdn switch-type primary-niisdn incoming-voice modemisdn guard-timer 3000isdn T203 10000isdn T306 30000isdn T310 4000isdn disconnect-cause 1fair-queue 64 256 0no cdp enableinterface Serial1:23no ip addressisdn switch-type primary-niisdn incoming-voice modemisdn guard-timer 3000isdn T203 10000isdn disconnect-cause 1fair-queue 64 256 0no cdp enable!interface FastEthernet0ip address 10.1.1.1 255.255.255.0no ip route-cacheno ip mroute-cacheduplex autospeed autoip rsvp bandwidth 7 7!ip classlessip route 10.0.0.0 255.0.0.0 172.18.193.1ip route 172.18.0.0 255.255.0.0 172.18.193.1no ip http serverip pim bidir-enable!ip radius source-interface Ethernet0!map-class dialer testdialer voice-calldialer-list 1 protocol ip permit!radius-server host 172.18.192.108 auth-port 1645 acct-port 1646radius-server retransmit 1radius-server key labradius-server vsa send accountingradius-server vsa send authenticationcall rsvp-synccall application voice voice_billing tftp://172.18.207.16/app_passport_silent.2.0.0.0.tcl!voice-port 0:Dvoice-port 1:Dvoice-port 3:0!no mgcp timer receive-rtcp!mgcp profile default!dial-peer voice 10 potsdestination-pattern 2021010119port 3:0prefix 2021010119!dial-peer voice 11 potsincoming called-number 3111100destination-pattern 3100802progress_ind progress enable 8port 0:Dprefix 93100802!dial-peer voice 36 voipapplication sessionincoming called-number 3100802destination-pattern 3100801session protocol sipv2session target ipv4:172.18.193.100codec g726r16!dial-peer voice 5 voipdestination-pattern 5555555session protocol sipv2session target ipv4:172.18.192.218!dial-peer voice 12 potsdestination-pattern 3111100prefix 93111100!dial-peer voice 19 potsdestination-pattern 2017030200port 1:Dprefix 2017030200!dial-peer voice 30 voipdestination-pattern 36602voice-class codec 2session protocol sipv2session target ipv4:172.18.193.120dial-peer voice 47 potsdestination-pattern 2021030100port 3:0!dial-peer voice 3111200 potsdestination-pattern 311200prefix 93100802!dial-peer voice 31 voipdestination-pattern 36601session protocol sipv2session target ipv4:172.18.193.98!dial-peer voice 1234 voipincoming called-number 1234destination-pattern 1234session target loopback:rtp!gatewaytimer receive-rtcp 5!sip-uaaaa username proxy-authretry invite 1retry bye 1!line con 0exec-timeout 0 0line aux 0line vty 0 4password zebra!endCommand Reference
This section documents new commands. All other commands used with this feature are documented in the Cisco IOS Release12.2 command reference publications.
ip rtcp report interval
To configure the average reporting interval between subsequent Real-Time Control Protocol (RTCP) report transmissions, use the ip rtcp report interval command in global configuration mode. To restore the default value, use the no form of this command.
ip rtcp report interval value
no ip rtcp report interval
Syntax Description
value
Sets the average interval for RTCP report transmissions in milliseconds. The valid range is from 1 to 65,535.
Defaults
The default is 5000 milliseconds.
Command Modes
Global configuration
Command History
Usage Guidelines
The ip rtcp report interval command configures the average interval between subsequent RTCP report transmissions for a given voice session. For example, if the value argument is set to 25,000 milliseconds, then an RTCP report is sent every 25 seconds, on average.
For more information about RTCP, see RFC 1889, RTP: A Transport Protocol for Real-Time Applications.
Examples
The following example shows the ip rtcp report interval command interval being set to 5000 milliseconds:
Router(config)# ip rtcp report interval 5000Related Commands
timer receive-rtcp (SIP)
To enable the Real-Time Control Protocol (RTCP) timer and to configure a multiplication factor for the RTCP timer interval for the Session Initiation Protocol (SIP), use the timer receive-rtcp command in gateway configuration mode. To restore the default value, use the no form of this command.
timer receive-rtcp timer
no timer receive-rtcp
Syntax Description
timer
Sets multiples of the RTCP report transmission interval. The valid range is from 2 to 1000. The default is 5.
Defaults
The default multiplication factor is 5.
Command Modes
Gateway configuration
Command History
Usage Guidelines
When the ip rtcp report interval and timer receive-rtcp commands are configured, the gateway uses RTCP report detection, rather than Real-Time Protocol (RTP) packet detection, to determine whether calls on the gateway are still active or should be disconnected. This method is more reliable because there are periods during voice calls when one or both parties are not sending RTP packets.
One common example of a voice session in which no RTP is sent is when a caller dials into a conference call and mutes his endpoint. If voice activity detection (VAD, also known as silence suppression) is enabled, no RTP packets are sent while the endpoint is muted. However, the muted endpoint continues to send RTCP reports at the interval specified by the ip rtcp report interval command.
The timer receive-rtcp value argument (or Mfactor) is multiplied with the interval that is set using the ip rtcp report interval command. If no RTCP packets are received in the resulting time period, the call is disconnected. The gateway signals the disconnect to the SIP network and the TDM network so that upstream and downstream devices can clear their resources. The gateway sends a SIP BYE to disconnect the call and sends a Q.931 DISCONNECT back to the TDM network to clear the call upon the expiration of the timer. The Q.931 DISCONNECT is sent with a cause code value of 3 (no route). There is no Q.931 Progress Indicator (PI) value included in the DISCONNECT.
To show timer-related output for SIP calls, use the debug ccsip events command.
Examples
The following example shows the multiplication factor being set to10 (or x * 10, where x is the interval that is set with the ip rtcp report interval command):
Router(config)# gatewayRouter(config-gateway)# timer receive-rtcp 10Router(config-gateway)# exitRelated Commands
Glossary
call—In SIP, a call consists of all participants in a conference invited by a common source. A SIP call is identified by a globally unique call identifier. A point-to-point IP telephony conversation maps into a single SIP call.
DSP—digital signal processor. Specialized computer chip designed to perform speedy and complex operations on digitized waveforms. Useful in processing sound, such as voice phone calls, and video.
Q.931—ITU-T Recommendation for signaling to establish, maintain, and clear ISDN network connections. Recommendation for specifying the UNI signaling protocol in N-ISDN. Q.931 was developed for out-of-band call control.
QoS—quality of service. Measure of performance for a transmission system that reflects its transmission quality and service availability.
RTCP—Real-Time Control Protocol. Monitors the QoS of an IPv6 RTP connection and conveys information about the ongoing session.
RTP—Real-Time Transport Protocol. A network protocol used to carry packetized audio and video traffic over an IP network.
session—A SIP session is a set of multimedia senders and receivers and the data streams flowing between the senders and receivers. A SIP multimedia conference is an example of a session. The called party can be invited several times by different calls to the same session.
SIP—Session Initiation Protocol. An application-layer protocol originally developed by the Multiparty Multimedia Session Control (MMUSIC) working group of the Internet Engineering Task Force (IETF). Their goal was to equip platforms to signal the setup of voice and multimedia calls over IP networks. SIP features are compliant with IETF RFC 2543, published in March 1999.
TDM—time-division multiplexing. A technique for transmitting a number of separate data, voice, and video signals simultaneously over one communications medium by quickly interleaving a piece of each signal one after the other.
VAD—voice activity detection. When enabled on voice port or a dial peer, silence is not transmitted over the network, only audible speech. When VAD is enabled, the sound quality is slightly degraded, but the connection monopolizes much less bandwidth.
