[Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?

Anthony Minessale anthony.minessale at gmail.com
Tue Apr 28 08:30:03 PDT 2009


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 <
mikael at bjerkeland.com> 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:21651019 at domain.appsvrslip11.prigw.com<sip%3A21651019 at domain.appsvrslip11.prigw.com>SIP/2.0
>   Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK
>   Route: <sip:21651019 at 1.1.1.1 <sip%3A21651019 at 1.1.1.1>>
>   Max-Forwards: 69
>   From: "someone" <sip:23695000 at 2.2.2.2 <sip%3A23695000 at 2.2.2.2>
> >;tag=m2SepeSZ63e3g
>   To: <sip:21651019 at domain.appsvrslip11.prigw.com<sip%3A21651019 at domain.appsvrslip11.prigw.com>
> >
>   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
>   CSeq: 114345142 INVITE
>   Contact: <sip:mod_sofia at 2.2.2.2:5060>
>   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:23695000 at 2.2.2.2<sip%3A23695000 at 2.2.2.2>
> >
>
>   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-eth1/21651019 at domain.appsvrslip11.prigw.com;fs_path=
> sip:21651019 at 1.1.1.1 <sip%3A21651019 at 1.1.1.1> 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:23695000 at 2.2.2.2 <sip%3A23695000 at 2.2.2.2>
> >;tag=m2SepeSZ63e3g
>   To: <sip:21651019 at domain.appsvrslip11.prigw.com<sip%3A21651019 at domain.appsvrslip11.prigw.com>
> >
>   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:23695000 at 2.2.2.2 <sip%3A23695000 at 2.2.2.2>
> >;tag=m2SepeSZ63e3g
>   To:
> <sip:21651019 at domain.appsvrslip11.prigw.com<sip%3A21651019 at domain.appsvrslip11.prigw.com>
> >;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=-
>   e=unknown at invalid.net
>   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-eth1/21651019 at domain.appsvrslip11.prigw.com;fs_path=
> sip:21651019 at 1.1.1.1 <sip%3A21651019 at 1.1.1.1> 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=-
> e=unknown at invalid.net
> 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-eth1/21651019 at domain.appsvrslip11.prigw.com;fs_path=
> sip:21651019 at 1.1.1.1 <sip%3A21651019 at 1.1.1.1> 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-eth1/21651019 at domain.appsvrslip11.prigw.com;fs_path=
> sip:21651019 at 1.1.1.1 <sip%3A21651019 at 1.1.1.1>] 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-eth1/21651019 at domain.appsvrslip11.prigw.com;fs_path=
> sip:21651019 at 1.1.1.1 <sip%3A21651019 at 1.1.1.1>!
> 2009-04-28 15:55:58 [DEBUG] switch_channel.c:1736
> switch_channel_perform_mark_pre_answered() Send signal
> sofia/internal/mikael-nokia at fs.voip.domain [BREAK]
> 2009-04-28 15:55:58 [DEBUG] switch_ivr_originate.c:1972
> switch_ivr_originate() sofia/internal/mikael-nokia at fs.voip.domain
> 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/mikael-nokia at fs.voip.domain
> 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/mikael-nokia at fs.voip.domain 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/mikael-nokia at fs.voip.domain] 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/mikael-nokia at fs.voip.domain!
> 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
> > <mikael at bjerkeland.com> 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
> >         Freeswitch-users at lists.freeswitch.org
> >         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: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>
> > pstn:213-799-1400
> > _______________________________________________
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>
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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