<br><br><div><span class="gmail_quote">On 8/8/06, <b class="gmail_sendername">Chan Kwang Mien</b> <<a href="mailto:kwangmien@asgent-tech.com">kwangmien@asgent-tech.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<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.</font></div></blockquote><div><br>it can - the only problem is that it needs to do transcoding and since g.729 is proprietary and the owner wants some royalty payments from it then you are stuck in the mud
<br><br><br><br> </div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><span class="q"><font color="#800000" face="Courier New, Courier">
> 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 color="#0000ff" face="Courier New, Courier"><u><a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com
</a><br>
</u></font><font color="#800000" face="Courier New, Courier">>
[<a href="mailto:asterisk-users-bounces@lists.digium.com%5DOn" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">mailto:</a></font><font color="#0000ff" face="Courier New, Courier"><u><a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
asterisk-users-bounces@lists.digium.com</a></u></font><a href="mailto:asterisk-users-bounces@lists.digium.com%5DOn" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"><font color="#800000" face="Courier New, Courier">
]On</font></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></span></div><div><span class="e" id="q_10cee3fdd8bbfb7f_2">
> On 8/8/06, Chan Kwang Mien <
<font color="#0000ff" face="Courier New, Courier"><u><a href="mailto:kwangmien@asgent-tech.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">kwangmien@asgent-tech.com</a></u></font></span></div>
<div><font color="#800000" face="Courier New, Courier">>
wrote:</font></div><font color="#800000" face="Courier New, Courier"></font><div><span class="e" id="q_10cee3fdd8bbfb7f_4"><font color="#800000" face="Courier New, Courier"><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></font></span></div><div><span class="q">
<font color="#800000" face="Courier New, Courier">> know how I could make them compatible
?<br>
><br>
><br></font></span></div><div><span class="q">
<font color="#800000" face="Courier New, Courier">> Thank you.<br>
><br>
> Regards,<br>
> Kwang Mien<br>
><br>
><br>
><br>
>
_______________________________________________<br></font></span></div><div><span class="q">
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