probably pstn side has acknowledged our gsm then sent ulaw anyway and we think its gsm.<br>most likely there are multiple codecs in the accept packet from the gateway and they expect us to figure out what codec to use based on the first packet we get from them rather than just accepting one codec in the sdp like 90% of devices so we have a proper chance to setup optimal packetization. This is one of those lame parts of the RFC that describe complete unscalable stupidity that some stuff likes to tout for who knows why.<br>
<br>one thing you can try is to set the variable aboslute_codec_string in the dial to force<br>only gsm to be advertised at all making it impossible for the remote end to respond with multiple codecs.<br><br><action application="bridge" data="{absolute_codec_string=GSM}sofia/<profile>/<uri>"/><br>
<br><br><br><br><div class="gmail_quote">On Fri, Nov 28, 2008 at 6:36 PM, Maxim Karp <span dir="ltr"><<a href="mailto:mkarp@securesilence.com">mkarp@securesilence.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello,<br>
<br>
I am using a GSM based endpoint connected to freeswitch that makes calls to<br>
the PSTN via a SIP gateway (SBC). The SBC uses PCMU between itself and<br>
freeswitch.<br>
<br>
When I make an outgoing call from a GSM based device via freewsitch to the<br>
PSTN via the SBC, everything works fine and audio works in both directions<br>
for both end points. I looked at the console logs and they do indicate that<br>
I am using GSM.<br>
<br>
Console output when I dial and before answer on the GSM device:<br>
<br>
v=0<br>
o=- 74 0 IN IP4 <a href="http://10.229.0.58" target="_blank">10.229.0.58</a><br>
s=session<br>
c=IN IP4 <a href="http://10.229.0.58" target="_blank">10.229.0.58</a><br>
b=CT:17<br>
t=0 0<br>
m=audio 59806 RTP/AVP 8 0 3 97 101<br>
k=base64:P6l1kBQy3canYTWZkxccjAVtTWO9g/N5L4gxLtX0UnM<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:97 RED/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=encryption:optional<br>
<br>
Console output once it rings and after I answer on the PSTN side:<br>
<br>
v=0<br>
o=FreeSWITCH 1227887572 1227887573 IN IP4 <a href="http://10.229.0.10" target="_blank">10.229.0.10</a><br>
s=FreeSWITCH<br>
c=IN IP4 <a href="http://10.229.0.10" target="_blank">10.229.0.10</a><br>
t=0 0<br>
a=sendrecv<br>
m=audio 30896 RTP/AVP 3 101 13<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=rtpmap:13 CN/8000<br>
a=ptime:20<br>
<br>
When I receive a call from the SIP gateway, the endpoint making the call<br>
(not on freeswitch) can't hear me speaking from the GSM device connected to<br>
freeswitch. I can hear everything fine on the GSM device.<br>
<br>
Here is the console output for the call info coming in from the PSTN.<br>
<br>
v=0<br>
o=FreeSWITCH 1227902084 1227902085 IN IP4 <a href="http://38.113.164.132" target="_blank">38.113.164.132</a><br>
s=FreeSWITCH<br>
c=IN IP4 <a href="http://38.113.164.132" target="_blank">38.113.164.132</a><br>
t=0 0<br>
a=sendrecv<br>
m=audio 16724 RTP/AVP 0 101 13<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=rtpmap:13 CN/8000<br>
a=ptime:20<br>
<br>
Here is how I have vars.xml configured:<br>
<br>
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/><br>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/><br>
<br>
<br>
When I prioritize GSM on the outbound codec prefs I get static on the PSTN<br>
side.<br>
<br>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA "/><br>
<br>
Any ideas?<br>
<br>
Maxim.<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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