[Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !!

[Steven Critchfield]

I think your problem comes from a misunderstanding of how the calls are
placed. With your canreinvite=no in the ATA section, you end up with the
ATA negotiating with asterisk for a call leg. Then you have asterisk
negotiating for the other call leg. Since the RTP stream is going
through asterisk, it behaves with how asterisk is capable. If there had
been a reinvite, then the ATA and the remote end would then be able to
negotiate a different codec.



On Mon, 2004-01-05 at 01:29, SamW wrote:
> Hello,
> 
> I have been trying to get my coders to work without a conversion. I have 
> read all the available asterisk documentation and support groups without 
> any luck. Here is my issue. (Please feel free to ask questions if you do 
> not understand what I am talking about.)
> 
> I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if 
> sip-server request g711)
> 
> I have 2 SIP-services to which I have to deliver the call in 2 coder 
> formats. Lets call 2 sip-providers, SIP-A and SIP-B. SIP-A accept g729 and 
> g711, SIP-B only accept g711.
> 
> I do not have any g729 licence, but I believe the * should negotiate to 
> have the correct passthrough coders as ATA is capable of both coders. (I 
> think even if you have the licenses, * should try avoid codec-conversions 
> when ever it can)
> 
> 
> Here is my settings in sip.conf. I will only list the required codec 
> related lines, for easy understanding,
> 
> [general]
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> 
> register => sip-a at foo.com
> register => sip-b at bar.com
> 
> [sip-a]
> ....
> disallow=all
> allow=ulaw
> 
> 
> [sip-b]
> ...
> disallow=all
> allow=g729
> 
> [ATA]
> .....
> canreinvite=no
> 
> Here is what happens when I look at the SIP packets from linux. (ethereal)
> 
> 
> Case 1 : ATA Dialing out through sip-a
> ============================
> 
> 
> ATA indicate that it can have following, codecs in SDP packet, in following 
> order
> ATA --> asterisk  INVITE message
> 	g729
> 	ulaw
> 	alaw
> asterisk  --> sip-a INVITE message (Note that already the order of coders 
> are changed. Is this how it should be I do not know. And how * decide what 
> order of coders to send to sip-a)
> 	alaw
> 	ulaw
> 	g729
> sip-a --> asterisk Session Progress Message
> 	ulaw
> asterisk --> ATA Session in Progress Message
> 	ulaw
> 	alaw
> 	g729
> asterisk --> ATA send a BYE message and hang up.
> 
> at this point asterisk indicate it cannot native bridge message. I do not 
> know why asterisk behaves like this, and I do think if asterisk send the 
> message back to ATA with g729 in its message it should have worked fith 
> nating bridging.
> 
> WARNING[1248642112]: File channel.c, Line 1853 
> (ast_channel_make_compatible): No path to translate from 
> SIP/sip-a-1e15(256) to SIP/4097-96d8(4)
> 
> 	
> 
> Case 2 : ATA calling sip-b
> ===================
> 	
> ATA indicate that it can have following, codecs in SDP packet, in following 
> order
> ATA --> asterisk  INVITE message
> 	g729
> 	ulaw
> 	alaw
> asterisk  --> sip-b INVITE message (Note that unlike case 1, the decision 
> by * in this case is OK. * only send one available coder info to the sip-b, 
> which is correct as per my config)
> 	ulaw
> 
> sip-b --> asterisk Session Progress Message
> 	ulaw
> asterisk --> ATA Session in Progress Message (Here again * sending multiple 
> choices to the ATA, I expect this to be only one request as * already know 
> from sip-b, that sip-b can only do ulaw. * know from 2 ways here one from 
> Session Progress message above and other from sip-b context that sip-b can 
> only do ulaw.) I am confused ????
> 	ulaw
> 	alaw
> 	g729
> 
> Asterisk send a BYE message to sip-b and send a 403 Forbidden Message to 
> ATA and hang-up the call here.
> 
> asterisk --> sip-a send a BYE message and hang up.
> asterisk -> ATA 403 Forbidden
> 
> NOTICE[1248642112]: File channel.c, Line 1478 (ast_set_read_format): Unable 
> to find a path from G729A to ULAW
> NOTICE[1248642112]: File channel.c, Line 1448 (ast_set_write_format): 
> Unable to find a path from ULAW to G729A
> 
> 
> =====================================================
> 
> Summery ;
> Why this is happening this way, (Do I not understand how to configure or is 
> this a bug?)
> As the coder negotiation is not well documented anywhere can you please 
> help me figure out how to configure the coder negotion.
> 
> IMHO, I belive that for each context, we need to have a way to force which 
> coder to choose.   True that * can code convert with license, but when you 
> code/decode it will always be lossy and will loose quality of sound.  If 
> one side is fixed for a particular codec, and the other side is flexible 
> for a negotiation, I should see that flexible side should get adjusted to 
> the correct codec. It do not seem to happen.
> 
> 
> Thank you in advance and appreciate your help.
> 
> 
> - Sam 
> 
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-- 
Steven Critchfield <critch at basesys.com>