Steve,
My Problem is not a problem, with the codec negotiation between end points. But when asterisk does it with canreinvite=no, * do not do it right. I replied with a lengthy discussion about my findings here, This behavior can be reproduced. But '*' do not seem to do the negotiation correctly.
http://lists.digium.com/pipermail/asterisk-users/2004-January/032197.html
>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 => [EMAIL PROTECTED]
> register => [EMAIL PROTECTED]
>
> [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 <[EMAIL PROTECTED]>
- SamW
