[asterisk-users] Problems with Codecs in Asterisk

[Rosli Sukri]

On 8/8/06, Chan Kwang Mien <kwangmien at asgent-tech.com> wrote:
>
> 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.
>

it can - the only problem is that it needs to do transcoding and since
g.729is proprietary and the owner wants some royalty payments from it
then you
are stuck in the mud





> 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%5DOn>*
> asterisk-users-bounces at lists.digium.com*]On<asterisk-users-bounces at lists.digium.com%5DOn>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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