we are facing a behavior with SIP third Party devices on cucm 8.6.2aSU3
The behavior is that in every call the device use the callmanager as MTP to place the call and the RTP stream is always through the callmanager.
We would like that the RTP goes directly between the devices as happens with Cisco IP Phones.
Maybe the two SIP devices do not have compatible codecs so they are going through an MTP to comply with the audio capabilities? You should try and check SDP message exchange between the two using RTMT Traces.
Also make sure that within the SIP Profile the flag "Media Termination Point Required" is not checked.
we made some test and the problem is only in calls to PSTN.
internal calls works fine with RTP between endpoints.
The Voice Gateway is connected in H.323 with Callmanager.
Yes, a cisco 2921 with IOS: c2900-universalk9-mz.SPA.152-4.M2.bin
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
These codecs are applied in the dial peers.
no it is not checked.
I tried configuring a SIP trunk to the Voice Gateway and a Dedicated Route Pattern to my mobile number to make a test.
And the RTP streaming goes directly between the SIP third party device and the Voice Gateway.
The problem is if you have a customer with hundreds of H323 Voice Gateway is not so good to change all the architecture setting up SIP trunks in all sites.
Anyone is able to get direct RTP between the SIP third party device and the Voice Gateway using H323 protocol?
The MTP is most probably being alocated due to a DTMF mismatch. The third party SIP phone is using Inband DTMF but an H.323 gateway can only to Out of band DTMF with the CUCM. Thus we need an MTP to achieve this.
thank you for the information.
So is it impossible to use third party with H323 gateway correctly?
Isn't there any workaround in your opinion?