Troubleshooting SIP Trunks with Cisco CallManager 5.x and Later

Transcription

1 Troubleshooting SIP Trunks with Cisco CallManager 5.x and Later Document ID: Contents Introduction Prerequisites Requirements Components Used Conventions SIP with CallManager 5.x Media Termination Point for SIP Trunks Call Drops and Cross Talk on SIP Trunks SIP and DTMF Signaling DTMF Signaling from Cisco MGCP Gateways to the IVR Services with SIP Trunks Outbound Calls from an IP Phone Through a SIP Trunk to the PSTN Fails with a Fast Busy Tone SIP Trunks Overview on CUCM CUCM not transmitting DTMF across SIP trunk Problem SIP Trunk Problem with Fast Busy Tone Media Termination Points Related Information Introduction The RFC 2833 Dual Tone Multifrequency (DTMF) Media Termination Point (MTP) Passthrough feature passes DTMF tones transparently between Session Initiation Protocol (SIP) endpoints that require either transcoding or use of the Resource Reservation Protocol (RSVP) Agent feature. Note: Cisco Unified CallManager 5.0 introduced major enhancements for SIP trunks. Prerequisites Requirements Cisco recommends that you have knowledge of these topics: Session Initiation Protocol (SIP), refer to Understanding Session Initiation Protocol (SIP). Cisco CallManager 5.x administration Components Used The information in this document is based on these software and hardware versions:

2 Cisco CallManager 5.x and later Cisco IOS SIP trunk The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command. Conventions Refer to Cisco Technical Tips Conventions for more information on document conventions. SIP with CallManager 5.x Cisco Unified CallManager 5.0 introduced major enhancements for SIP trunks and overcame the limitations of earlier releases of Cisco Unified CallManager, such as single codec support, lack of video support, and the mandatory media termination point (MTP) for RFC 2833 DTMF support. The other enhancements to SIP trunks in Cisco Unified CallManager 5.0 are the support for REFER, header replacement, Subscribe/Notify, message waiting indication (MWI), MTP removal, video support, multiple SIP trunks per inbound port number, SIP Redirection 3XX, transport layer security (TLS), digest authentication, call preservation, and T.38 fax relay. Media Termination Point for SIP Trunks You can configure Cisco CallManager SIP devices (lines and trunks) to always use an MTP. If the configuration parameters are not set to use an MTP (default case), Cisco CallManager attempts to dynamically allocate an MTP if the DTMF methods for calls that are not compatible. For example, SCCP phones support only out of band DTMF, and Cisco SIP phones (model 7905, 7912, 7940, 7960) support RFC2833. Because the DTMF methods are not identical, Cisco CallManager dynamically allocates an MTP. If a SCCP phone that supports RFC2833 and out of band, such as Cisco IP phone 7971, calls a Cisco SIP IP phone 7940, Cisco CallManager does not allocate an MTP because both phones support RFC2833. With the same type of DTMF method supported on each phone, there is no need for an MTP. Call Drops and Cross Talk on SIP Trunks Cisco Unified Communications Manager 5.x is connected to a Cisco IOS voice gateway with a SIP trunk as shown. PSTN calls to IP phones registered with Cisco Unified Communications Manager 5.0 drop intermittently. This also results in cross talk in both inbound, as well as outbound calls.

3 SIP and DTMF Signaling The DTMF information is transported between SIP endpoints with out of band (OOB) and in band signaling. In band DTMF transport methods send DTMF tones as either raw tones in the RTP media stream or as signaled tones in the RTP payload with RFC Among SIP product vendors, RFC 2833 has become the predominant method to send and receive DTMF tones and is supported by the majority of Cisco voice products. Endpoints can negotiate the use of RFC 2833 or an out of band DTMF method end to end but, if a common DTMF method cannot be negotiated between the endpoints, Cisco Unified Communications Manager 5.x inserts an MTP dynamically. Note: In Cisco Unified Communications Manager 5.x, the number of concurrent SIP calls is limited by the number of MTP resources available. In Cisco Unified Communications Manager 4.x, all SIP trunks are required to allocate an MTP for DTMF and Early Offer support. With Cisco Unified Communications Manager 5.x, the Media Termination Point Required check box is disabled by default. If you remove this restriction, it increases the overall performance and frees up the MTP resources for other applications to use. Endpoints can negotiate the use of RFC 2833 or an out of band DTMF method end to end but, if a common DTMF method cannot be negotiated between the endpoints, Cisco Unified Communications Manager 5.x inserts an MTP dynamically. In order to fix this issue, uncheck Media Termination Point Required under the SIP Trunk configuration in the Cisco Unified Communications Manager 5.x administration page so that Cisco Unified Communications Manager 5.x allocates MTP resources dynamically, dependent upon the end points. DTMF Signaling from Cisco MGCP Gateways to the IVR Services with SIP Trunks DTMF signaling is not passed from the Cisco voice gateways (that run MGCP to CallManager) to the IVR services (that run SIP Trunks to CallManager) without the enablement of MTP required. After enabling the require MTP option in CallManager, this issue is resolved because the MTP is able to translate the out of band DTMF signaling in MGCP to the in band signaling that is used for the SIP trunks. Unfortunately, there are limits on the number of software MTPs that can be enabled on CallManager, which means that the solution does not scale as required: one MTP is required per call. If the MTP requirement can be moved to the Cisco voice gateways, more DSP resources are required. It is better to simply pass the DTMF tones within the media stream straight to the IVR servers, without any out of band DTMF tones. This also provides the added benefit of not passing DTMF tones in the MGCP signaling back to CallManager, a potential security issue when customer account numbers / pins are used. In order to enable RFC 2833 for MGCP gateway with CCM 5.0 SIP trunk, without the requirement of enabling MTP, configure these two commands in the MGCP gateway: mgcp dtmf relay voip codec all mode nte ca mgcp package capability fm package Note: These commands are included in the Cisco IOS 12.4T IP VOICE feature set and later.

4 Outbound Calls from an IP Phone Through a SIP Trunk to the PSTN Fails with a Fast Busy Tone When an off net call is attempted through the Session Initiation Protocol (SIP) trunk, the caller gets a fast busy tone. All inbound calls to the IP phones are fine. In order to resolve the issue, follow these steps. 1. Verify whether or not there are enough media resources available. 2. Verify whether or not the Cisco IP Voice Media Streaming Service has started. If it has not, restart the service. 3. Verify whether or not the proper codecs are used. Then, reset the Software Media Termination Point (MTP) in the Cisco CallManager. 4. Check the protocol configured under System > Security Profile > SIP Trunk Security Profile for outbound and inbound calls. By default, the Cisco CallManager tries Transport Control Protocol (TCP) on outbound calls. This can time out if it is not able to establish a TCP connection with the SIP gateway. If the protocol is set to TCP, change it to User Datagram Protocol (UDP). SIP Trunks Overview on CUCM A SIP trunk configuration is used to set up communication with a SIP User Agent such as another Cisco Unified Communications Manager cluster or a SIP gateway. Today, SIP is arguably the most commonly chosen protocol when connecting to service providers and Unified Communications applications. Cisco Unified Communications Manager 8.5 and later releases provide these SIP trunk and call routing enhancements: Can run on all Unified Communications Manager nodes Up to 16 destination IP addresses per trunk SIP OPTIONS ping keepalives SIP Early Offer support for voice and video calls (insert MTP if needed) QSIG over SIP SIP trunk normalization and transparency Supports the use of route lists on all Unified Communications Manager nodes CUCM not transmitting DTMF across SIP trunk Problem DTMF does not work on Cisco 7936 conference phones. Topology: SIP provider< SIP CUBE< SIP< CUCM< CUCM 7.1< SCCP CP7936 Cisco 7936 conference units are configured to route out a CUBE to a SIP provider. In this case, the issue is that DTMF is not sent out. Voic and WebEx sessions cannot be used.

5 The reason the SIP trunk has DTMF issues only for CP7936 is due to hardware limitations of the phone. The phones do not support RTP NTE. In order to resolve this issue, you must have a Media Termination Point (MTP) on CUCM to convert out of band DTMF to RTP NTE for the SIP provider. In this case, you must either use a 7937 model or enable MTP or HW Transcoder so that the out of band DTMF sent by the 7936 is converted to NTE. The 7937 does not demonstrate the same problem since it supports NTE. SIP Trunk Problem with Fast Busy Tone A fast busy tone is received when a call is made from one CallManager cluster to another. In this case, you must collect the CUCM traces of the timestamp in order to analyze them. The traces show that the codec that is negotiated is G.711 because the MTP box is checked and the preferred codec is set as G.711 (under the SIP trunk configuration in the administration page of CUCM) even though the Region used by the phone and SIP trunk are set to use G.729 instead. In addition, it shows that when the 183 Session Progress is received by CUCM the codec is accepted and the call moves to connect status. Media Termination Points Media Termination Points (MTPs) are generally not required for Delayed Offer calls from Unified CallManager SIP trunks. For this reason, Cisco recommends Delayed Offer as the call setup method for outbound calls from Unified CallManager SIP trunks. For outbound Early Offer calls from Unified CallManager, MTP resources are required (SIP Trunk MTP required box is checked) and remain in the media path for the duration of the call. For calls inbound and outbound from Unified CallManager, endpoints can negotiate the use of RFC 2833 or an out of band DTMF method (for example, KPML) end to end. If a common DTMF method cannot be negotiated between the endpoints, Cisco Unified Communications Manger 5.x and later releases will insert an MTP dynamically. Cisco Unified Communications Manger 5.x and later releases support Delayed Offer (Invite without SDP) by default. Although Delayed Offer is a mandatory part of the SIP RFC 3261 specification, some SIP applications do not support it. In those cases, you must configure the SIP trunk to support Early Offer by pre allocating an MTP under the SIP trunk configuration. MTPs are available in three forms: Software based MTPs in Cisco IOS gatewaysavailable with any Cisco IOS T train software release and scaling up to 500 sessions (calls) on the Cisco 3845 Integrated Services Router. Hardware based MTPs in Cisco IOS gatewaysavailable with any Cisco IOS T train software release; hardware MTPs use on board DSP resources and scale calls according to the number of DSPs supported on the Cisco router platform. Software based MTPs using the Cisco IP Voice Media Streaming Application on a Cisco Media Convergence Server (MCS). This example configuration is for a Cisco IOS software based MTP:! sccp local Vlan5 sccp ccm identifier 5 version 5.0.1

6 ! Communications Manager IP address ( ) sccp! sccp ccm group 5 bind interface Vlan5 associate ccm 5 priority 1 associate profile 5 register MTP000E83783C50! MTP name (MTP000E83783C50)... must match the Unified CM MTP name.! dspfarm profile 5 mtp description software MTP codec g711ulaw codec pass through maximum sessions software 500 associate application SCCP Related Information Failover Timer on SIP Trunks with CallManager Configuration Example Best Practices When Implementing SIP Trunks for PSTN Access Cisco IOS SIP Configuration Guide Cisco Unified CallManager Administration Guide Technical Support & Documentation Cisco Systems Contacts & Feedback Help Site Map Cisco Systems, Inc. All rights reserved. Terms & Conditions Privacy Statement Cookie Policy Trademarks of Cisco Systems, Inc. Updated: Jul 21, 2011 Document ID: 98524