CUCM SIP Trunk with SONUS, MTP or not MTP ?

[MBO46]

Hi All,

I work on a SIP trunk between a Call Manager 11.0 and a SONUS 1000,

I have an issue with MOH, without MTP i have a blank and with MTP it's good.

In the traces i can see that without MTP, the Call Manager asks to the SONUS to connect to the IP address 0.0.0.0 for the MOH.

Why the call manager don't connect the SONUS on his IPVMS service ?

I don't want to use MTP because i don't want that all STREAM pass-trough the CallManager.

Best regards,

Matthieu

[Manish Gogna]

Hi Matthieu,

You can enable the following parameter on the SIP profile " Early Offer support for voice and video calls Mandatory (insert MTP if needed)" instead of enabling MTP on the SIP trunk itself.

When you configure Early Offer support for voice and video calls Mandatory (insert MTP if needed) on the SIP Profile of a trunk, calls from older SCCP-based phones, SIP Delayed Offer trunks, and H.323 Slow Start trunks cause Unified CM to allocate an MTP, if an MTP or transcoder is not already allocated for that call for another reason. The MTP is used to generate an Offer SDP with a valid media port number and IP address. The MTP is allocated from the media resources that are associated with the calling device rather than from the media resources of the outbound SIP trunk. (This prevents the media path from being anchored to the MTP of the outbound SIP trunk.) If the MTP cannot be allocated from the media resource group list (MRGL) of the calling device, the MTP allocation is attempted from the MRGL of the SIP trunk.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/media.html#pgfId-1231293

Manish

[MBO46]

I have enable the option "Early Offer support for voice and video calls Mandatory (insert MTP if needed)"  on the SIP profile but without MTP checked on the SIP trunk the MTP resource is not assigned.

[Mohammed al Baqari]

Hi,

From CUCM RTMT, can you send Real-Time Session Trace snapshot for the call while placing it on hold.

When you resume the call, is it going as expected, i.e connecting back to holder?

As Vivek mentioned, 0.0.0.0 is expected as this is used by CUCM to disable the current media stream along with attribute a=inactive.

[MBO46]

With NO MTP on SIP trunk, MOH Unicast is NOT used. (at the beginning of the graph)

With MTP on SIP trunk, MOH Unicast is used (at the end of the graph)

[Mohammed al Baqari]

Hi,

If MoH resource isn't allocated without MTP, then phase 1 of MoH process is failing which is stopping the current media stream.

When CUCM sends INVITE with 0.0.0.0, is it receiving 200OK from SONUS?

Also, please confirm if resume is working.

[MBO46]

You can see the call flow just below,

[Vivek Batra]

Hi Mat,

Would you like to check couple of points when MTP is disabled;

1. Put the call on hold and see the output of 'show perf query class "Cisco MOH Device" and observe the output whether unicast source for this call is assigned or not.

2. Which codec is enabled for MoH in IPVMS service parameter?

3. How does region (/device pool) settings look between MoH server and SIP trunk? It is set to 8 or 64 Kbps?

- Vivek

[MBO46]

Hi Vivek,

Now, with the option "Early Offer support for voice and video calls Mandatory (insert MTP if needed)" on the SIP profile, i have a unicast stream toward the SONUS in the correct CODEC. But the MOH don't arrive to the external Phone.

I think this is the SONUS with the SIP negotiation which don't forward the stream.

I will troubleshoot the SONUS side,

thanks for your help !

[Vivek Batra]

Hi,

Thanks for the update.

- Vivek

[MBO46]

You have right Mohammed al Baqari, i looked in the traces and i found a bad codec negotiation with the MOH_2 (MOH software on the CUCM)

Now, without MTP on the trunk, when PhoneA put PhoneB on hold, i can see now in RTMT that a MOH resource is used, but i have always a blank and not the default Cisco MOH :(

[Mohammed al Baqari]

Good to see that part one is resolved. 

Now what codec are you using between MoH and phoneB.

Also, check the reachability between phoneB and MoH server. 

Since this is working with MTP, I am assuming that user hold audio source configured on phoneA is present on the MoH server.

[Vivek Batra]

Hi Mat,

That is one of the way to put call in hold, old RFC specified to use ip as 0.0.0.0 in sdp body for one way audio. New RFC changes this attribute to sendonly.

For CUCM to play MOH, it does not need to include its IP in sdp body of re-invite. Can you please check MOH server, mrg, mrgl config is in place and reference to correct SIP trunk.

If configuration seems fine, you can check from CUCM CLI about status of MOH unicast source and share the result.

[MBO46]

Hi,

Thanks for your answers,

Call Flow :

PhoneA=====CUCM===[SIP trunk]===SONUS 1000==[SIP Trunk]==IPBX===PhoneB

So, Without MTP on SIP TRUNK, When PhoneA put PhoneB onhold > NO MOH and i have this,

screen_shot_01-29-16_at_09.48_am.png

With MTP Check on SIP TRUNK, when PhoneA put PhoneB onhold > MOH ok and i have this,

screen_shot_01-29-16_at_09.53_am.png

Ressource MOH is OK on the SIP trunk

[MBO46]

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