Are you geting 183+sdp from the nokia?<br>the media timer only operates once media is established and only <br>counts against you if the channel is being read from and that does not <br>happen until you get a 183 or 200 w/sdp<br>
<br>try putting a debug line in switch_rtp.c around 1520<br>printf(&quot;MISSED PACKETS %u/%u\n&quot;, rtp_session-&gt;missed_count, rtp_session-&gt;max_missed_packets);  <br><br>but try updating first there was a recent fix that may have prevented a timer surge at the beginning of calls.<br>
<br><br><div class="gmail_quote">On Tue, Apr 28, 2009 at 6:20 AM, Mikael Aleksander Bjerkeland <span dir="ltr">&lt;<a href="mailto:mikael@bjerkeland.com">mikael@bjerkeland.com</a>&gt;</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;">
Hi,<br>
<br>
I have been testing inbound calls to a Nokia phone with handover to a<br>
cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and<br>
had to set rtp-timeout to a very low 6 seconds in order to get &quot;fast&quot;<br>
handover. This introduces an interesting side-effect that hangs up calls<br>
even in the ringing state after 6 seconds. Is this the desired behaviour<br>
of rtp-timeout-sec? My initial guess was that rtp-timeout-sec should<br>
only be valid for established calls where the two endpoints have<br>
exchanged rtp at some point but have stopped exchanging media. As far as<br>
I know a phone call in ringing state has not shared any RTP with the<br>
other endpoint until it gets early media or is answered. Should<br>
rtp-timeout-sec really be valid even when ringing?<br>
<br>
It seems to me that setting rtp-timeout-sec to 60 seconds would add an<br>
absolute time limit on ringing phone calls to 60 seconds, which I<br>
believe is not the actual purpose of this limit. Could anyone please<br>
share their thoughts on this matter?<br>
<br>
<br>
Thanks,<br>
Mikael<br>
<br>
<br>
<br>
<br>
<br>
_______________________________________________<br>
Freeswitch-users mailing list<br>
<a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
</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>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400<br>