[asterisk-dev] asterisk 16 / pjsip: SDP handling on outbound calls - unnecessary transcoding

[Joshua C. Colp]

On Sat, Jun 22, 2019, at 4:40 PM, Michael Maier wrote:
> Hello!
> 
> On outbound calls, the codecs for the subsequent INVITE to the ISP are 
> derived from the codecs defined for the trunk and from the received 
> INVITE from the local device.
> 
> The codecs presented by the ISP in 200 OK seem to be ignored 
> unfortunately later on.
> 
> E.g.:
> direct_media				disabled
> Local endpoint definition: 		g722, alaw, ulaw
> Trunk definition: 			g722, alaw, ulaw
> 
> Outbound call:
> Initial INVITE from local endpoint: 	g722, alaw, ulaw, g726
> 
> Subsequent INVITE to ISP: 		g722, alaw, ulaw (the codecs are derived 
> from the initial INVITE and the codecs defined for the trunk)
> Response from ISP: 200 OK: 		alaw
> 
> 200 OK to local endpoint: 		g722, alaw, ulaw (based on initial INVITE 
> and local endpoint definition - the 200 OK from the ISP is ignored)
> 
> => asterisk has to transcode between g722 <-> alaw
> 
> 
> Is there any chance to get the codec information of the ISP during 
> building the 200 OK for the local endpoint in 
> handle_negotiated_sdp_session_media() or maybe somewhere else? Or did I 
> miss something else? I want to remove g722 in 200 OK to
> local endpoint.

There's nothing built in explicitly to do this and it's not something that has really been scoped out. A few people have wanted such functionality, but it hasn't been implemented. You can't really reach across and get the information - it has to be communicated back probably through using a frame.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org