[Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?
anthony.minessale at gmail.com
Tue Apr 28 05:50:38 PDT 2009
Are you geting 183+sdp from the nokia?
the media timer only operates once media is established and only
counts against you if the channel is being read from and that does not
happen until you get a 183 or 200 w/sdp
try putting a debug line in switch_rtp.c around 1520
printf("MISSED PACKETS %u/%u\n", rtp_session->missed_count,
but try updating first there was a recent fix that may have prevented a
timer surge at the beginning of calls.
On Tue, Apr 28, 2009 at 6:20 AM, Mikael Aleksander Bjerkeland <
mikael at bjerkeland.com> wrote:
> I have been testing inbound calls to a Nokia phone with handover to a
> cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and
> had to set rtp-timeout to a very low 6 seconds in order to get "fast"
> handover. This introduces an interesting side-effect that hangs up calls
> even in the ringing state after 6 seconds. Is this the desired behaviour
> of rtp-timeout-sec? My initial guess was that rtp-timeout-sec should
> only be valid for established calls where the two endpoints have
> exchanged rtp at some point but have stopped exchanging media. As far as
> I know a phone call in ringing state has not shared any RTP with the
> other endpoint until it gets early media or is answered. Should
> rtp-timeout-sec really be valid even when ringing?
> It seems to me that setting rtp-timeout-sec to 60 seconds would add an
> absolute time limit on ringing phone calls to 60 seconds, which I
> believe is not the actual purpose of this limit. Could anyone please
> share their thoughts on this matter?
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
Anthony Minessale II
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the FreeSWITCH-users