as soon as FS sees 183 it expects media. if they send 183 and no media it will most certainly timeout
On Tue, Apr 28, 2009 at 9:33 AM, Mikael Aleksander Bjerkeland < [email protected]> wrote: > The scenario I was referring to was actually an outbound call from a > locally registered SIP phone to a cellphone. The same thing happens > whether I use a SIP or PRI trunk. After 6 s it hangs up. > > > I get SDP on 183 no matter whether I'm calling a cellphone or a fixed > line. I also get ringing indication. The 183+sdp is passed on to the > Nokia and after 6 s the call is hung up. Both the SIP and PRI trunks > claim to send early media but there seems to be no audio/RTP. If I > answer the call in 6 s it's not dropped because the media path was > established before RTP timeout. > > The same thing happens on latest trunk. > I added the debug line at 1520 and did make && /etc/init.d/freeswitch > stop && make install && /etc/init.d/freeswitch start but the debug line > didn't show up anywhere in the CLI. > > Is my upstream provider doing something wrong in sending early media in > these cases? Seems pretty odd. It can be avoided by setting a higher > rtp-timeout-sec but it will still be an absolute timeout on ringing. > > > A transcript of the log: > > send 1293 bytes to udp/[1.1.1.1]:5060 at 13:55:56.451865: > > ------------------------------------------------------------------------ > INVITE > sip:[email protected]<sip%[email protected]>SIP/2.0 > Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK > Route: <sip:[email protected] <sip%[email protected]>> > Max-Forwards: 69 > From: "someone" <sip:[email protected] <sip%[email protected]> > >;tag=m2SepeSZ63e3g > To: > <sip:[email protected]<sip%[email protected]> > > > Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc > CSeq: 114345142 INVITE > Contact: <sip:[email protected]:5060> > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13175M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 383 > P-Asserted-Identity: "someone" <sip:[email protected]<sip%[email protected]> > > > > v=0 > o=FreeSWITCH 3718974841365302606 4309079514688066219 IN IP4 2.2.2.2 > s=FreeSWITCH > c=IN IP4 2.2.2.2 > t=0 0 > m=audio 52706 RTP/AVP 9 8 0 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > m=video 52752 RTP/AVP 99 > a=rtpmap:99 H264/90000 > > ------------------------------------------------------------------------ > 2009-04-28 15:55:56 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() > Channel > sofia/external-eth1/[email protected];fs_path= > sip:[email protected] <sip%[email protected]> entering state [calling][0] > recv 305 bytes from udp/[1.1.1.1]:5060 at 13:55:56.482864: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > From: "someone"<sip:[email protected] <sip%[email protected]> > >;tag=m2SepeSZ63e3g > To: > <sip:[email protected]<sip%[email protected]> > > > Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc > CSeq: 114345142 INVITE > Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 1035 bytes from udp/[1.1.1.1]:5060 at 13:55:58.296906: > > ------------------------------------------------------------------------ > SIP/2.0 183 Session Progress > From: "someone"<sip:[email protected] <sip%[email protected]> > >;tag=m2SepeSZ63e3g > To: > <sip:[email protected]<sip%[email protected]> > >;tag=20134330840200942815366 > Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc > CSeq: 114345142 INVITE > Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK > content-type: application/sdp > contact: <sip:1.1.1.1:5060;nt_end_pt=YM0 > > +~K!-.f0vfc830~P68.cio~H9zwgW0VyisWTdcaM26c610Xbo1.nfS.5NQt3mO~~70!-.f0vft815;nt_server_host=1.1.1.1> > supported: 100rel > x-nt-party-id: -/ > allow: ACK > allow: BYE > allow: CANCEL > allow: INVITE > allow: OPTIONS > allow: INFO > allow: SUBSCRIBE > allow: REFER > allow: NOTIFY > allow: PRACK > server: CS2000_NGSS/9.0 > Content-Length: 300 > > v=0 > o=IWSPM 573585738 573585738 IN IP4 84.20.97.100 > s=- > [email protected] > t=0 0 > m=audio 45954 RTP/AVP 8 0 18 101 > c=IN IP4 84.20.97.100 > a=ptime:20 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > m=video 0 RTP/AVP 99 > c=IN IP4 2.2.2.2 > a=rtpmap:99 H264/90000 > > ------------------------------------------------------------------------ > 2009-04-28 15:55:58 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() > Channel > sofia/external-eth1/[email protected];fs_path= > sip:[email protected] <sip%[email protected]> entering state > [proceeding][183] > 2009-04-28 15:55:58 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() > Remote SDP: > v=0 > o=IWSPM 573585738 573585738 IN IP4 84.20.97.100 > s=- > [email protected] > t=0 0 > m=audio 45954 RTP/AVP 8 0 18 101 > c=IN IP4 84.20.97.100 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > m=video 0 RTP/AVP 99 > c=IN IP4 2.2.2.2 > a=rtpmap:99 H264/90000 > > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20] > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:1912 > sofia_glue_tech_set_codec() Set Codec > sofia/external-eth1/[email protected];fs_path= > sip:[email protected] <sip%[email protected]> PCMA/8000 20 ms 160 samples > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() > AUDIO RTP > [sofia/external-eth1/[email protected];fs_path= > sip:[email protected] <sip%[email protected]>] 2.2.2.2 port 52706 -> > 84.20.97.100 port 45954 codec: 8 ms: 20 > 2009-04-28 15:55:58 [DEBUG] switch_rtp.c:906 switch_rtp_create() > Starting timer [soft] 160 bytes per 20ms > 2009-04-28 15:55:58 [NOTICE] sofia_glue.c:2573 sofia_glue_tech_media() > Pre-Answer > sofia/external-eth1/[email protected];fs_path= > sip:[email protected] <sip%[email protected]>! > 2009-04-28 15:55:58 [DEBUG] switch_channel.c:1736 > switch_channel_perform_mark_pre_answered() Send signal > sofia/internal/[email protected] [BREAK] > 2009-04-28 15:55:58 [DEBUG] switch_ivr_originate.c:1972 > switch_ivr_originate() sofia/internal/[email protected] > receive message [PROGRESS] > 2009-04-28 15:55:58 [INFO] mod_sofia.c:1377 sofia_receive_message() > Asked to send early media by sofia/internal/[email protected] > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20] > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:1912 > sofia_glue_tech_set_codec() Set Codec > sofia/internal/[email protected] PCMA/8000 20 ms 160 samples > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 98 > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() > AUDIO RTP [sofia/internal/[email protected]] 10.100.4.192 port > 58072 -> 10.247.3.253 port 49152 codec: 8 ms: 20 > 2009-04-28 15:55:58 [DEBUG] switch_rtp.c:906 switch_rtp_create() > Starting timer [soft] 160 bytes per 20ms > 2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2325 sofia_glue_activate_rtp() > Set comfort noise payload to 13 > 2009-04-28 15:55:58 [NOTICE] sofia_glue.c:2573 sofia_glue_tech_media() > Pre-Answer sofia/internal/[email protected]! > 2009-04-28 15:55:58 [INFO] mod_sofia.c:1420 sofia_receive_message() Ring > SDP: > v=0 > o=FreeSWITCH 1240868886 1240868887 IN IP4 10.100.4.192 > s=FreeSWITCH > c=IN IP4 10.100.4.192 > t=0 0 > m=audio 58072 RTP/AVP 8 98 13 > a=rtpmap:8 PCMA/8000 > a=rtpmap:98 telephone-event/8000 > a=fmtp:98 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > a=sendrecv > > > > El mar, 28-04-2009 a las 07:50 -0500, Anthony Minessale escribió: > > 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, > > rtp_session->max_missed_packets); > > > > 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 > > <[email protected]> wrote: > > Hi, > > > > 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? > > > > > > Thanks, > > Mikael > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > [email protected] > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:[email protected] <msn%[email protected]> > > GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:[email protected] <sip%[email protected]> > > iax:[email protected]/888 > > googletalk:[email protected]<googletalk%3aconf%[email protected]> > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > [email protected] > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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