Hi All,<br><br>I am trying to get mod_dingaling working with gtalk.<br>I have configured extention 1004 in jingle_profiles/client.xml.<br>I make a call from another gtalk user,the call lands up fine at the extention I have<br>
configured.But there is no audio.I searched a lot and have tried everything I found <br>on the net including using different STUN servers, using different values for candidate-acl,disable-rtp-auto-adjust. etc.<br><br> I enabled debugging in mod_dingaling and found in the xmpp messages that along with a <br>
local RTP ,and a stun RTP a relay RTP port is being advertized by the google server,I ran wireshark on my freeswitch server and looks like <br>the audio is going to the stun RTP port not to the realy RTP port.<br>Can this be a problem?<br>
<br>Does gtalk want the audio to be sent directly to the concerned gtalk client or does it have to go through the google server?<br>The FreeSwitch vesrion I am using is ' FreeSWITCH version: 1.0.7 (svn-exported)'.<br>
<br><br>Please find below my jingle_profiles/client.xml.<br><br> <include><br> <!-- Client Profile (Original mode) --><br> <!-- to use this profile take the x- away from the open and close tags so its <profile> and </profile> --><br>
<profile type="client"><br> <param name="name" value="jclient"/><br> <param name="login" value="<a href="http://XXXXXXXX@gmail.com/gtalk">XXXXXXXX@gmail.com/gtalk</a>"/><br>
<param name="password" value="XXXXX"/><br> <param name="dialplan" value="XML"/><br> <param name="context" value="public"/><br> <param name="message" value="Jingle all the way"/><br>
<param name="rtp-ip" value="$${bind_server_ip}"/><br> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/><br> <param name="auto-login" value="true"/><br>
<!-- SASL "plain" or "md5" --><br> <param name="sasl" value="plain"/><br> <!-- if the server where the jabber is hosted is not the same as the one in the jid --><br>
<param name="server" value="<a href="http://talk.google.com">talk.google.com</a>"/><br> <!-- Enable TLS or not --><br> <param name="tls" value="true"/><br>
<!-- disable to trade async for more calls --><br> <param name="use-rtp-timer" value="true"/><br> <param name="disable-rtp-auto-adjust" value="true"/><br>
<br> <!-- default extension (if one cannot be determined) --><br> <param name="exten" value="1004"/><br> <!-- VAD choose one --><br> <!-- <param name="vad" value="in"/> --><br>
<!-- <param name="vad" value="out"/> --><br> <param name="vad" value="both"/><br> <!--<param name="avatar" value="/path/to/tiny.jpg"/>--><br>
<param name="candidate-acl" value="wan.auto"/><br> <param name="local-network-acl" value="localnet.auto"/><br> </profile><br></include><br><br><br>I will appreciate any help or pointers on this.<br>
<br>--<br>Thanks<br>Saurabh<br><br><br>