<div dir="ltr"><br><div><div>Hi,<br></div><div><br>I&#39;m having some difficulties making freeswitch talk to gmail chat.<br></div>Signalling and text chat work both ways, but have troubles with audio. I&#39;m behind NAT and use STUN to traverse it.<br>

<br></div>In a problematic case I observe the following error in debug 
mode in the console, where user1 initiated the conversation with user2<br><div><br><br>&lt;iq to=&quot;user1@XXX/fsvega0EC80FEA&quot; id=&quot;313&quot; type=&quot;error&quot; from=&quot;user2@YYY/VtokiPhn75120521&quot;&gt;<br>

  &lt;ses:session type=&quot;candidates&quot; id=&quot;3759110558&quot; initiator=&quot;user1@XXX/fsvega0EC80FEA&quot; xmlns:ses=&quot;<a href="http://www.google.com/session" target="_blank">http://www.google.com/session</a>&quot;&gt;<br>

    &lt;ses:candidate name=&quot;rtp&quot; address=&quot;xxx.xxx.xxx.xxx&quot; port=&quot;23751&quot; 
username=&quot;bSYbFfMybRj8D3xH&quot; password=&quot;ri5AOommjwum60f8&quot; preference=&quot;1.0&quot;
 protocol=&quot;udp&quot; type=&quot;stun&quot; network=&quot;0&quot; generation=&quot;0&quot;&gt;&lt;/ses:candidate&gt;<br>
    &lt;ses:candidate name=&quot;rtcp&quot; address=&quot;xxx.xxx.xxx.xxx&quot; port=&quot;23752&quot;
 username=&quot;tVI0bVcGkAl7kdQT&quot; password=&quot;ay23gfVzUPdaKDW4&quot; 
preference=&quot;1.0&quot; protocol=&quot;udp&quot; type=&quot;stun&quot; network=&quot;0&quot; 
generation=&quot;0&quot;&gt;&lt;/ses:candidate&gt;<br>
  &lt;/ses:session&gt;<br>  &lt;error type=&quot;modify&quot;&gt;<br>    &lt;sta:bad-request xmlns:sta=&quot;urn:ietf:params:xml:ns:xmpp-stanzas&quot;&gt;&lt;/sta:bad-request&gt;<br>    &lt;sta:text xml:lang=&quot;en&quot; xmlns:sta=&quot;urn:ietf:params:xml:ns:xmpp-stanzas&quot;&gt;&lt;/sta:text&gt;<br>

  &lt;/error&gt;<br>&lt;/iq&gt;<br><br></div><div>It appears that user2 
would like to modify the session, but mod_dingaling does not take any 
action and I do not see any RTP traffic with wireshark. Is this 
expected?<br><br>
</div><div>A second question is: In the session setup log I see 
mod_dingaling advertising a port which is outside the range specified 
with the configuration variables rtp-start-port and rtp-end-port in switch.xml. Do the
 latter only work with SIP?<br>
<br></div><div>Thanks for any advise,<br></div>Fede</div>