Thanks! I'll notify them of the problem and see if there's a way around it.
2009/4/28 Anthony Minessale <anthony.miness...@gmail.com>
> 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
>> > _______________________________________________
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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_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
>
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