I was wondering if anybody has encountered the following problem:
The PROBLEM is that if the codec negotiated between Phone A and Phone B is different than the codec negotiated between Phone B and the SIP trunk to the recording server, Phone B does not send any audio streams to the recording server.
After continuous testing we found that if the codecs are the same the stream gets sent, otherwise it does not.
After viewing the trace files I saw that CUCM locks the codec negotiated between Phone A and Phone B, and after that it sees that the call between Phone B and the recording server requires a different codec. It tries to allocate transcoding resources but it does not succeed.
I tried allocating the software MTPs as well as hardware ones provided by some DSP modules on local voice gateways.
It did not work.
I found that somebody on the forum posted the same issue but in regard to the Transfer functionality of an IP Phone with call recording configured (the problem was caused by the codec mismatch).
No responses to this thread, and we are looking at the same problem. I am not able to find any changes in the documentation or solutions on-line. From the Features and Services Guide:
The codecs for recording calls match the codec of the customer-agent call.
There doesn't seem to be any change from 7.0 to 7.1 to 8.0.
Has anyone found a workaround other than allowing g.711 over teh WAN or forcing every call in the enterprise that could possibly be recorded to g.729?
After seeing that nobody replyed to my post I tried to redo the configuration again, more carefully this time.
I did the following:
1. Configure a Hardware transcoder for each Location/Device Pool using the hardware DSPs located in the local voice gateway from each location
2. Configure a Media Resource Group List and Media Resource Group List for each Location/Device Pool containing the previously configured Hardware Transcoders
2. Attach the MRGL to the respective IP Phones and also to the SIP Trunks pointing to the Recording Server (CallREC in my case)
3. Check the "Media Termination Point Required" checkbox in the SIP Trunk configuration page
4. Tested the recording again with every scenario that I could think of and it worked in every situation:
This is aone situation that did not work before:
Before it did not work because I forgot to check the "MTP Required" checkbox.
The HW Transcoder is required in order to transcode to G.729. Simply configuring a HW transcoder, attaching it to the devices and not checking "MTP Required" checkbox does not solve the problem (this is where i did the mistake previously).
Hope this helps.
If not, let me know, maybe we can find a solution .
I set up transcoders using dragos instructions and found that when the MTP required setting is checked that 4 sessions are used when transcoding is needed and 2 if transcoding isn't needed. 2 for the required MTP and 2 for transcoding if needed. I removed the MTP required setting and transcoding still worked and only used 2 sessions for the transcoding and wasn't used at all if transcoding wasn't needed.
Since CUCM 7.0 use g.722 as a default codec, my voce recorder from RedBox was not recording internal calls (calls that were made with g.722 codec).
Here are the steps that I used to solve the issue:
Use steps 1-4 from dragos.frantz and add the following commands to
dspfarm profile 2 transcode
Hope this will solve the issue.