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 <
mik...@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:21651...@domain.appsvrslip11.prigw.com<sip%3a21651...@domain.appsvrslip11.prigw.com>SIP/2.0
> Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK
> Route: <sip:21651...@1.1.1.1 <sip%3a21651...@1.1.1.1>>
> Max-Forwards: 69
> From: "someone" <sip:23695...@2.2.2.2 <sip%3a23695...@2.2.2.2>
> >;tag=m2SepeSZ63e3g
> To:
> <sip:21651...@domain.appsvrslip11.prigw.com<sip%3a21651...@domain.appsvrslip11.prigw.com>
> >
> Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
> CSeq: 114345142 INVITE
> Contact: <sip:mod_so...@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:23695...@2.2.2.2<sip%3a23695...@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-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=
> sip:21651...@1.1.1.1 <sip%3a21651...@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:23695...@2.2.2.2 <sip%3a23695...@2.2.2.2>
> >;tag=m2SepeSZ63e3g
> To:
> <sip:21651...@domain.appsvrslip11.prigw.com<sip%3a21651...@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:23695...@2.2.2.2 <sip%3a23695...@2.2.2.2>
> >;tag=m2SepeSZ63e3g
> To:
> <sip:21651...@domain.appsvrslip11.prigw.com<sip%3a21651...@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=unkn...@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-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=
> sip:21651...@1.1.1.1 <sip%3a21651...@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=unkn...@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-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=
> sip:21651...@1.1.1.1 <sip%3a21651...@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-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=
> sip:21651...@1.1.1.1 <sip%3a21651...@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-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=
> sip:21651...@1.1.1.1 <sip%3a21651...@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-no...@fs.voip.domain [BREAK]
> 2009-04-28 15:55:58 [DEBUG] switch_ivr_originate.c:1972
> switch_ivr_originate() sofia/internal/mikael-no...@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-no...@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-no...@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-no...@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-no...@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
> > <mik...@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@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_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org>
> > iax:gu...@conference.freeswitch.org/888
> > googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org>
> > pstn:213-799-1400
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
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> http://www.freeswitch.org
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org>
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org>
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