[asterisk-users] Problems with Codecs in Asterisk

[Chan Kwang Mien]

 From the SIP messages exchange, sip1 informs Asterisk in the INVITE
message that it supports g.729 and g.711u. Asterisk then compares its
first allowed codec which is g.729 with the supported codec by sip1. Since
sip1 supports g.729 and it is an "allowed" codec, Asterisk chooses g.729
as the codec between itself and sip1.

Asterisk then forwards the INVITE message but the codec in the INVITE is
changed to g.711u. sip3 replied that it supports g.711u in the OK message.
Asterisk then realised that the codec between itself and sip3 is different
from the codec between itself and sip1. There is a need for transcoding.
And since there isn't any g.729 Licence, the connection breaks.

In short, once Asterisk is sure that the first codec of the allowed list
is supported by sip1, it will use that codec and will ignore the remaining
codec, in this case, g.711u.

Intuitively, I thought that since sip1 supports both g.729 and g.711u, it
should be able to connect to a g.729 phone or a g.711u phone via Asterisk
using the same sip.conf.


 > I have the same problem here, why does asterisk not use ulaw with Sip1 ->
 > Sip3 ?  As it has allow=g729 and allow=ulaw in Sip1, should it not
 > fallback onto ulaw when the g729 fails?
 >
 > Thanks,
 > Dean.
 >   -----Original Message-----
 >   From: asterisk-users-bounces at lists.digium.com
 > [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Rosli Sukri
 >   Sent: 08 August 2006 13:38
 >   To: Asterisk Users Mailing List - Non-Commercial Discussion
 >   Subject: Re: [asterisk-users] Problems with Codecs in Asterisk
 >
 >
 >   either
 >   1)pay digium for g.729 license or
 >   2)allow g.729 for sip3
 >
 >   - sip 1 -> sip2 work cause it will pass thru,
 >   - sip 2 -> sip3 fails because since asterisk wants to do transcoding to
 > 729<->711 and no license
 >   if bandwidth is a concern just use GSM (if available as a codec on the
 > phone)
 >
 >
 >   On 8/8/06, Chan Kwang Mien < kwangmien at asgent-tech.com> wrote:
 >     Hi,
 >
 >     My test-setup is as follows :
 >
 >     sip1 <--> Asterisk <--> sip2
 >                   ^
 >                   |-------> sip3
 >
 >     In sip.conf,
 >
 >     [sip1]
 >     type=friend
 >     host=dynamic
 >     secret=pass
 >     disallow=all
 >     allow=g729
 >     allow=ulaw
 >
 >     [sip2]
 >     type=friend
 >     host=dynamic
 >     secret=pass
 >     disallow=all
 >     allow=g729
 >
 >     [sip3]
 >     type=friend
 >     host=dynamic
 >     secret=pass
 >     disallow=all
 >     allow=ulaw
 >
 >
 >     sip1 supports g.729 and g.711u only
 >     sip2 supports g.729 only
 >     sip3 supports g.711u only
 >
 >     sip1 is able to establish a call to sip2.
 >     However, I have problem establishing a call from sip1 to sip3. sip3
 >     rings but when I answered it, it hanged up.
 >
 >     The Logs are :
 >
 >         -- Executing Dial("SIP/2006-389a", "SIP/2003") in new stack
 >         -- Called 2003
 >     Aug  8 09:55:15 WARNING[6937]: channel.c:2725
 >     ast_channel_make_compatible: No path to translate from
 > SIP/2003-b5f8(4)
 >     to SIP/2006-389a(256)
 >
 >         -- SIP/2003-b5f8 is ringing
 >         -- SIP/2003-b5f8 answered SIP/2006-389a
 >
 >     Aug  8 09:55:16 WARNING[6937]: channel.c:2725
 >     ast_channel_make_compatible: No path to translate from
 >     SIP/2006-389a(256) to SIP/2003-b5f8(4)
 >     Aug  8 09:55:16 WARNING[6937]: app_dial.c:1608 dial_exec_full: Had to
 >     drop call because I couldn't make SIP/2006-389a compatible with
 >     SIP/2003-b5f8
 >       == Spawn extension (phones, 2003, 1) exited non-zero on
 >     'SIP/2006-389a'
 >
 >
 >     I think the codecs used by sip3 and sip1 are incompatible. Does anyone
 >     know how I could make them compatible ?
 >
 >
 >     Thank you.
 >
 >     Regards,
 >     Kwang Mien
 >
 >
 >
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