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MTP requirement for video calls in SIP Trunk

Fatih Cetin

Hello everybody,

In this scenario:

          IP Phone (video) ------ CUCM (7.1.X) ------SIP TRUNK ------- CUBE ------- SIP TRUNK ------- CUCM (7.1.X) ------ IP Phone (video)

If I select "Media Termination Point Required" check box in both SIP Trunk configuration, the signalling path would be:

          IP Phone - CUCM - CUBE - CUCM - IP Phone.

And the media path would be (considering the CUBE is set for media flow-through):

     IP Phone - CUCM - CUBE - CUCM - IP Phone. (Same as signalling path)

Is this assumption correct? If so, during a video call between two IP Phones, the media will be terminated and reoriginated in: first, in the first CUCM, then in CUBE, and finally in the second CUCM.

Would it be possible to have a video call in such a scenario?



5 Replies 5

Jagpreet Singh
Cisco Employee
Cisco Employee

Hi Faith,

In case you are using CUCM based MTPs, the video calls will not work. For that matter fax calls will also not work. This is because this media cannot be repackatized.

If you need to use an MTP, you can use an IOS basee MTP with the codec passthru command under the dspfarm profile.



Sent from Cisco Technical Support Android App

Adding to Jagpreet's post, the built in CUCM MTP can be used for video and fax as of CUCM 8.6 since codec pass-thru was added to the CUCM MTPs starting with that version.  Since you're on 7.1, Jagpreet is correct that an IOS MTP would be needed here.

Thank you very much Jagpreet and Joseph.

So I have to either upgrade the Call Manager to the version 8.6 (or higher) or use an IOS MTP.

Can you please tell me what would be the minimum requirements for IOS MTP? What kind of router shall I use (is 2800 series router OK?)? What should be the IOS version? Which PVDM shall I use?

Thank you very much,


The codec pass-through option was added to IOS in 12.4(4)T, so anything newer than that supports codec pass-through, as for the router a 2800 is ok with pvdm-2s for DSP resources.

Hi Joe,


I am facing the same issue. I need the same SIP trunk to do both audio and video calls from the telepresence room. The call flow is starting from the call manager and then going in through the SIP trunk to VCS.


When I am enabling require MTP on the trunk DTMF tones for the calls coming from the gateway works but the video calls from the telepresence using the same trunk goes to an audio only call. When I uncheck MTP on the trunk video call from the telepresence works but the DTMF tones stops getting relayed from the gateway to the SIP trunk to WebEx.


I tried using IOS MTP and configuring the codec pass through and then enabled MTP to see if it works but unfortunately ended up with the same result.


Your input is much appreciated.




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