<html>
<font face="Courier New, Courier">From the SIP messages exchange, sip1
informs Asterisk in the INVITE<br>
message that it supports g.729 and g.711u. Asterisk then compares
its<br>
first allowed codec which is g.729 with the supported codec by sip1.
Since<br>
sip1 supports g.729 and it is an "allowed" codec, Asterisk
chooses g.729<br>
as the codec between itself and sip1.<br>
<br>
Asterisk then forwards the INVITE message but the codec in the INVITE
is<br>
changed to g.711u. sip3 replied that it supports g.711u in the OK
message.<br>
Asterisk then realised that the codec between itself and sip3 is
different<br>
from the codec between itself and sip1. There is a need for
transcoding.<br>
And since there isn't any g.729 Licence, the connection breaks.<br>
<br>
In short, once Asterisk is sure that the first codec of the allowed list
<br>
is supported by sip1, it will use that codec and will ignore the
remaining<br>
codec, in this case, g.711u.<br>
<br>
Intuitively, I thought that since sip1 supports both g.729 and g.711u,
it<br>
should be able to connect to a g.729 phone or a g.711u phone via
Asterisk<br>
using the same sip.conf.<br>
<br>
<br>
</font><font face="Courier New, Courier" color="#800000">> I have the
same problem here, why does asterisk not use ulaw with Sip1 -><br>
> Sip3 ? As it has allow=g729 and allow=ulaw in Sip1, should it
not<br>
> fallback onto ulaw when the g729 fails?<br>
><br>
> Thanks,<br>
> Dean.<br>
> -----Original Message-----<br>
> From:
</font><font face="Courier New, Courier" color="#0000FF"><u>asterisk-users-bounces@lists.digium.com<br>
</u></font><font face="Courier New, Courier" color="#800000">>
[<a href="mailto:asterisk-users-bounces@lists.digium.com%5DOn" eudora="autourl">mailto:</a></font><font face="Courier New, Courier" color="#0000FF"><u>asterisk-users-bounces@lists.digium.com</u></font><a href="mailto:asterisk-users-bounces@lists.digium.com%5DOn" eudora="autourl"><font face="Courier New, Courier" color="#800000">]On</a>
Behalf Of Rosli Sukri<br>
> Sent: 08 August 2006 13:38<br>
> To: Asterisk Users Mailing List - Non-Commercial
Discussion<br>
> Subject: Re: [asterisk-users] Problems with Codecs in
Asterisk<br>
><br>
><br>
> either<br>
> 1)pay digium for g.729 license or<br>
> 2)allow g.729 for sip3<br>
><br>
> - sip 1 -> sip2 work cause it will pass thru,<br>
> - sip 2 -> sip3 fails because since asterisk wants to
do transcoding to<br>
> 729<->711 and no license<br>
> if bandwidth is a concern just use GSM (if available as
a codec on the<br>
> phone)<br>
><br>
><br>
> On 8/8/06, Chan Kwang Mien <
</font><font face="Courier New, Courier" color="#0000FF"><u>kwangmien@asgent-tech.com</u></font><font face="Courier New, Courier" color="#800000">>
wrote:<br>
> Hi,<br>
><br>
> My test-setup is as follows :<br>
><br>
> sip1 <--> Asterisk <-->
sip2<br>
>
^<br>
>
|-------> sip3<br>
><br>
> In sip.conf,<br>
><br>
> [sip1]<br>
> type=friend<br>
> host=dynamic<br>
> secret=pass<br>
> disallow=all<br>
> allow=g729<br>
> allow=ulaw<br>
><br>
> [sip2]<br>
> type=friend<br>
> host=dynamic<br>
> secret=pass<br>
> disallow=all<br>
> allow=g729<br>
><br>
> [sip3]<br>
> type=friend<br>
> host=dynamic<br>
> secret=pass<br>
> disallow=all<br>
> allow=ulaw<br>
><br>
><br>
> sip1 supports g.729 and g.711u only<br>
> sip2 supports g.729 only<br>
> sip3 supports g.711u only<br>
><br>
> sip1 is able to establish a call to
sip2.<br>
> However, I have problem establishing a call
from sip1 to sip3. sip3<br>
> rings but when I answered it, it hanged
up.<br>
><br>
> The Logs are :<br>
><br>
> -- Executing
Dial("SIP/2006-389a", "SIP/2003") in new stack<br>
> -- Called 2003<br>
> Aug 8 09:55:15 WARNING[6937]:
channel.c:2725<br>
> ast_channel_make_compatible: No path to
translate from<br>
> SIP/2003-b5f8(4)<br>
> to SIP/2006-389a(256)<br>
><br>
> -- SIP/2003-b5f8 is
ringing<br>
> -- SIP/2003-b5f8
answered SIP/2006-389a<br>
><br>
> Aug 8 09:55:16 WARNING[6937]:
channel.c:2725<br>
> ast_channel_make_compatible: No path to
translate from<br>
> SIP/2006-389a(256) to SIP/2003-b5f8(4)<br>
> Aug 8 09:55:16 WARNING[6937]:
app_dial.c:1608 dial_exec_full: Had to<br>
> drop call because I couldn't make
SIP/2006-389a compatible with<br>
> SIP/2003-b5f8<br>
> == Spawn extension (phones,
2003, 1) exited non-zero on<br>
> 'SIP/2006-389a'<br>
><br>
><br>
> I think the codecs used by sip3 and sip1 are
incompatible. Does anyone<br>
> know how I could make them compatible
?<br>
><br>
><br>
> Thank you.<br>
><br>
> Regards,<br>
> Kwang Mien<br>
><br>
><br>
><br>
>
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