Force only PCMA, PCMU in SDP to Gateway before bridge

[Skeelkat]

Is it possible to have FS force PCMA or PCMU only in the SDP before doing a Bridge to a gateway?

I have an upstream SBC that the moment any codec (regardless of order) is present in the SDP that it does not support, I get a Not Accepted here error. I have set the absolute_codec_string=PCMA,PCMA just before Bridge to Gateway in the outbound route but it does not seem to work.

 

[Adrian Fretwell]

Setting absolute_codec_string should work, this will fix the codec to what you have specified, nothing else will be added to the list.
Setting coded_string will establish your basic codec rules but allow other options to be added. For my specific requirements, I tend to do this globally in the dialplan rather than the outbound route:

 

[Skeelkat]

This is what I thought as well. or could deduce from the documentation

So this is effectively my outbound route (I will adapt the global dial plan if and when I get this to work as it should). I have specified a 4 as outbound digit. The INVITE Confirms that the upstream SBC is getting the number formatted correctly.




My SIP UA is set to use OPUS only (Internal Calls on SIP Internal Profile works 100% between OPUS / PCMA) and when I look at the INVITE to the Gateway, the SDP still shows this. IP Addresses blanked for security purposes. I need to have OPUS completely removed from this SDP.

Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): FreeSWITCH 1614659495 1614659496 IN IP4 xxx.xxx.xxx.xxx
Session Name (s): FreeSWITCH
Connection Information (c): IN IP4 xxx.xxx.xxx.xxx
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 28258 RTP/AVP 102 101
Media Attribute (a): rtpmap:102 opus/48000/2
Media Attribute (a): fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40
Media Attribute (a): rtpmap:101 telephone-event/48000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): silenceSuppff - - - -
Media Attribute (a): ptime:20
[Generated Call-ID: ce46ceb9-f5f4-1239-f19d-76ee32e17225]

This is driving me up the wall. I cannot get this to work for some reason. Any idea of Params that needs to be specified in the External SIP (Sofia) profile?

 

[KonradSC]

To force it just on the outbound b-leg to the provider you can use try this on the oubound route.

Action , Export , nolocal:absolute_codec_string=PCMU