[Freeswitch-users] mod_dingaling <error type="modify">

[Federico Beffa]

Hi,

I'm having some difficulties making freeswitch talk to gmail chat.
Signalling and text chat work both ways, but have troubles with audio. I'm
behind NAT and use STUN to traverse it.

In a problematic case I observe the following error in debug mode in the
console, where user1 initiated the conversation with user2


<iq to="user1 at XXX/fsvega0EC80FEA" id="313" type="error" from="user2 at YYY
/VtokiPhn75120521">
  <ses:session type="candidates" id="3759110558"
initiator="user1 at XXX/fsvega0EC80FEA"
xmlns:ses="http://www.google.com/session">
    <ses:candidate name="rtp" address="xxx.xxx.xxx.xxx" port="23751"
username="bSYbFfMybRj8D3xH" password="ri5AOommjwum60f8" preference="1.0"
protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="rtcp" address="xxx.xxx.xxx.xxx" port="23752"
username="tVI0bVcGkAl7kdQT" password="ay23gfVzUPdaKDW4" preference="1.0"
protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
  </ses:session>
  <error type="modify">
    <sta:bad-request
xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:bad-request>
    <sta:text xml:lang="en"
xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:text>
  </error>
</iq>

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?

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?

Thanks for any advise,
Fede
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130805/1653762e/attachment-0001.html