Thanks! I'll notify them of the problem and see if there's a way around it.<br><br><br><br><div class="gmail_quote">2009/4/28 Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">as soon as FS sees 183 it expects media.<br><br>if they send 183 and no media it will most certainly timeout <br>
<div><div></div><div class="h5"><br><div class="gmail_quote">On Tue, Apr 28, 2009 at 9:33 AM, Mikael Aleksander Bjerkeland <span dir="ltr"><<a href="mailto:mikael@bjerkeland.com" target="_blank">mikael@bjerkeland.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">The scenario I was referring to was actually an outbound call from a<br>
locally registered SIP phone to a cellphone. The same thing happens<br>
whether I use a SIP or PRI trunk. After 6 s it hangs up.<br>
<br>
<br>
I get SDP on 183 no matter whether I'm calling a cellphone or a fixed<br>
line. I also get ringing indication. The 183+sdp is passed on to the<br>
Nokia and after 6 s the call is hung up. Both the SIP and PRI trunks<br>
claim to send early media but there seems to be no audio/RTP. If I<br>
answer the call in 6 s it's not dropped because the media path was<br>
established before RTP timeout.<br>
<br>
The same thing happens on latest trunk.<br>
I added the debug line at 1520 and did make && /etc/init.d/freeswitch<br>
stop && make install && /etc/init.d/freeswitch start but the debug line<br>
didn't show up anywhere in the CLI.<br>
<br>
Is my upstream provider doing something wrong in sending early media in<br>
these cases? Seems pretty odd. It can be avoided by setting a higher<br>
rtp-timeout-sec but it will still be an absolute timeout on ringing.<br>
<br>
<br>
A transcript of the log:<br>
<br>
send 1293 bytes to udp/[1.1.1.1]:5060 at 13:55:56.451865:<br>
<br>
------------------------------------------------------------------------<br>
INVITE <a href="mailto:sip%3A21651019@domain.appsvrslip11.prigw.com" target="_blank">sip:21651019@domain.appsvrslip11.prigw.com</a> SIP/2.0<br>
Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK<br>
Route: <<a href="mailto:sip%3A21651019@1.1.1.1" target="_blank">sip:21651019@1.1.1.1</a>><br>
Max-Forwards: 69<br>
From: "someone" <<a href="mailto:sip%3A23695000@2.2.2.2" target="_blank">sip:23695000@2.2.2.2</a>>;tag=m2SepeSZ63e3g<br>
To: <<a href="mailto:sip%3A21651019@domain.appsvrslip11.prigw.com" target="_blank">sip:21651019@domain.appsvrslip11.prigw.com</a>><br>
Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc<br>
CSeq: 114345142 INVITE<br>
Contact: <<a href="http://sip:mod_sofia@2.2.2.2:5060" target="_blank">sip:mod_sofia@2.2.2.2:5060</a>><br>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13175M<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,<br>
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, presence, dialog, call-info, sla,<br>
include-session-description, presence.winfo, message-summary, refer<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 383<br>
P-Asserted-Identity: "someone" <<a href="mailto:sip%3A23695000@2.2.2.2" target="_blank">sip:23695000@2.2.2.2</a>><br>
<br>
v=0<br>
o=FreeSWITCH 3718974841365302606 4309079514688066219 IN IP4 2.2.2.2<br>
s=FreeSWITCH<br>
c=IN IP4 2.2.2.2<br>
t=0 0<br>
m=audio 52706 RTP/AVP 9 8 0 3 101 13<br>
a=rtpmap:9 G722/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=rtpmap:13 CN/8000<br>
a=ptime:20<br>
m=video 52752 RTP/AVP 99<br>
a=rtpmap:99 H264/90000<br>
<br>
------------------------------------------------------------------------<br>
2009-04-28 15:55:56 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()<br>
Channel<br>
sofia/external-eth1/<a href="mailto:21651019@domain.appsvrslip11.prigw.com" target="_blank">21651019@domain.appsvrslip11.prigw.com</a>;fs_path=<a href="mailto:sip%3A21651019@1.1.1.1" target="_blank">sip:21651019@1.1.1.1</a> entering state [calling][0]<br>
recv 305 bytes from udp/[1.1.1.1]:5060 at 13:55:56.482864:<br>
<br>
------------------------------------------------------------------------<br>
SIP/2.0 100 Trying<br>
From: "someone"<<a href="mailto:sip%3A23695000@2.2.2.2" target="_blank">sip:23695000@2.2.2.2</a>>;tag=m2SepeSZ63e3g<br>
To: <<a href="mailto:sip%3A21651019@domain.appsvrslip11.prigw.com" target="_blank">sip:21651019@domain.appsvrslip11.prigw.com</a>><br>
Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc<br>
CSeq: 114345142 INVITE<br>
Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK<br>
Content-Length: 0<br>
<br>
<br>
------------------------------------------------------------------------<br>
recv 1035 bytes from udp/[1.1.1.1]:5060 at 13:55:58.296906:<br>
<br>
------------------------------------------------------------------------<br>
SIP/2.0 183 Session Progress<br>
From: "someone"<<a href="mailto:sip%3A23695000@2.2.2.2" target="_blank">sip:23695000@2.2.2.2</a>>;tag=m2SepeSZ63e3g<br>
To:<br>
<<a href="mailto:sip%3A21651019@domain.appsvrslip11.prigw.com" target="_blank">sip:21651019@domain.appsvrslip11.prigw.com</a>>;tag=20134330840200942815366<br>
Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc<br>
CSeq: 114345142 INVITE<br>
Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK<br>
content-type: application/sdp<br>
contact: <sip:1.1.1.1:5060;nt_end_pt=YM0<br>
+~K!-.f0vfc830~P68.cio~H9zwgW0VyisWTdcaM26c610Xbo1.nfS.5NQt3mO~~70!-.f0vft815;nt_server_host=1.1.1.1><br>
supported: 100rel<br>
x-nt-party-id: -/<br>
allow: ACK<br>
allow: BYE<br>
allow: CANCEL<br>
allow: INVITE<br>
allow: OPTIONS<br>
allow: INFO<br>
allow: SUBSCRIBE<br>
allow: REFER<br>
allow: NOTIFY<br>
allow: PRACK<br>
server: CS2000_NGSS/9.0<br>
Content-Length: 300<br>
<br>
v=0<br>
o=IWSPM 573585738 573585738 IN IP4 84.20.97.100<br>
s=-<br>
e=<a href="mailto:unknown@invalid.net" target="_blank">unknown@invalid.net</a><br>
t=0 0<br>
m=audio 45954 RTP/AVP 8 0 18 101<br>
c=IN IP4 84.20.97.100<br>
a=ptime:20<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
m=video 0 RTP/AVP 99<br>
c=IN IP4 2.2.2.2<br>
a=rtpmap:99 H264/90000<br>
<br>
------------------------------------------------------------------------<br>
2009-04-28 15:55:58 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()<br>
Channel<br>
sofia/external-eth1/<a href="mailto:21651019@domain.appsvrslip11.prigw.com" target="_blank">21651019@domain.appsvrslip11.prigw.com</a>;fs_path=<a href="mailto:sip%3A21651019@1.1.1.1" target="_blank">sip:21651019@1.1.1.1</a> entering state [proceeding][183]<br>
2009-04-28 15:55:58 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state()<br>
Remote SDP:<br>
v=0<br>
o=IWSPM 573585738 573585738 IN IP4 84.20.97.100<br>
s=-<br>
e=<a href="mailto:unknown@invalid.net" target="_blank">unknown@invalid.net</a><br>
t=0 0<br>
m=audio 45954 RTP/AVP 8 0 18 101<br>
c=IN IP4 84.20.97.100<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=ptime:20<br>
m=video 0 RTP/AVP 99<br>
c=IN IP4 2.2.2.2<br>
a=rtpmap:99 H264/90000<br>
<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()<br>
Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20]<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()<br>
Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:1912<br>
sofia_glue_tech_set_codec() Set Codec<br>
sofia/external-eth1/<a href="mailto:21651019@domain.appsvrslip11.prigw.com" target="_blank">21651019@domain.appsvrslip11.prigw.com</a>;fs_path=<a href="mailto:sip%3A21651019@1.1.1.1" target="_blank">sip:21651019@1.1.1.1</a> PCMA/8000 20 ms 160 samples<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp()<br>
Set 2833 dtmf payload to 101<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp()<br>
AUDIO RTP<br>
[sofia/external-eth1/<a href="mailto:21651019@domain.appsvrslip11.prigw.com" target="_blank">21651019@domain.appsvrslip11.prigw.com</a>;fs_path=<a href="mailto:sip%3A21651019@1.1.1.1" target="_blank">sip:21651019@1.1.1.1</a>] 2.2.2.2 port 52706 -> 84.20.97.100 port 45954 codec: 8 ms: 20<br>
2009-04-28 15:55:58 [DEBUG] switch_rtp.c:906 switch_rtp_create()<br>
Starting timer [soft] 160 bytes per 20ms<br>
2009-04-28 15:55:58 [NOTICE] sofia_glue.c:2573 sofia_glue_tech_media()<br>
Pre-Answer<br>
sofia/external-eth1/<a href="mailto:21651019@domain.appsvrslip11.prigw.com" target="_blank">21651019@domain.appsvrslip11.prigw.com</a>;fs_path=<a href="mailto:sip%3A21651019@1.1.1.1" target="_blank">sip:21651019@1.1.1.1</a>!<br>
2009-04-28 15:55:58 [DEBUG] switch_channel.c:1736<br>
switch_channel_perform_mark_pre_answered() Send signal<br>
sofia/internal/mikael-nokia@fs.voip.domain [BREAK]<br>
2009-04-28 15:55:58 [DEBUG] switch_ivr_originate.c:1972<br>
switch_ivr_originate() sofia/internal/mikael-nokia@fs.voip.domain<br>
receive message [PROGRESS]<br>
2009-04-28 15:55:58 [INFO] mod_sofia.c:1377 sofia_receive_message()<br>
Asked to send early media by sofia/internal/mikael-nokia@fs.voip.domain<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()<br>
Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20]<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()<br>
Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:1912<br>
sofia_glue_tech_set_codec() Set Codec<br>
sofia/internal/mikael-nokia@fs.voip.domain PCMA/8000 20 ms 160 samples<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp()<br>
Set 2833 dtmf payload to 98<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp()<br>
AUDIO RTP [sofia/internal/mikael-nokia@fs.voip.domain] 10.100.4.192 port<br>
58072 -> 10.247.3.253 port 49152 codec: 8 ms: 20<br>
2009-04-28 15:55:58 [DEBUG] switch_rtp.c:906 switch_rtp_create()<br>
Starting timer [soft] 160 bytes per 20ms<br>
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2325 sofia_glue_activate_rtp()<br>
Set comfort noise payload to 13<br>
2009-04-28 15:55:58 [NOTICE] sofia_glue.c:2573 sofia_glue_tech_media()<br>
Pre-Answer sofia/internal/mikael-nokia@fs.voip.domain!<br>
2009-04-28 15:55:58 [INFO] mod_sofia.c:1420 sofia_receive_message() Ring<br>
SDP:<br>
v=0<br>
o=FreeSWITCH 1240868886 1240868887 IN IP4 10.100.4.192<br>
s=FreeSWITCH<br>
c=IN IP4 10.100.4.192<br>
t=0 0<br>
m=audio 58072 RTP/AVP 8 98 13<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:98 telephone-event/8000<br>
a=fmtp:98 0-16<br>
a=rtpmap:13 CN/8000<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
<br>
<br>
El mar, 28-04-2009 a las 07:50 -0500, Anthony Minessale escribió:<br>
<div><div></div><div>> Are you geting 183+sdp from the nokia?<br>
> the media timer only operates once media is established and only<br>
> counts against you if the channel is being read from and that does<br>
> not<br>
> happen until you get a 183 or 200 w/sdp<br>
><br>
> try putting a debug line in switch_rtp.c around 1520<br>
> printf("MISSED PACKETS %u/%u\n", rtp_session->missed_count,<br>
> rtp_session->max_missed_packets);<br>
><br>
> but try updating first there was a recent fix that may have prevented<br>
> a timer surge at the beginning of calls.<br>
><br>
><br>
> On Tue, Apr 28, 2009 at 6:20 AM, Mikael Aleksander Bjerkeland<br>
> <<a href="mailto:mikael@bjerkeland.com" target="_blank">mikael@bjerkeland.com</a>> wrote:<br>
> Hi,<br>
><br>
> I have been testing inbound calls to a Nokia phone with<br>
> handover to a<br>
> cellphone number if I get MEDIA_TIMEOUT on the B leg of the<br>
> call, and<br>
> had to set rtp-timeout to a very low 6 seconds in order to get<br>
> "fast"<br>
> handover. This introduces an interesting side-effect that<br>
> hangs up calls<br>
> even in the ringing state after 6 seconds. Is this the desired<br>
> behaviour<br>
> of rtp-timeout-sec? My initial guess was that rtp-timeout-sec<br>
> should<br>
> only be valid for established calls where the two endpoints<br>
> have<br>
> exchanged rtp at some point but have stopped exchanging media.<br>
> As far as<br>
> I know a phone call in ringing state has not shared any RTP<br>
> with the<br>
> other endpoint until it gets early media or is answered.<br>
> Should<br>
> rtp-timeout-sec really be valid even when ringing?<br>
><br>
> It seems to me that setting rtp-timeout-sec to 60 seconds<br>
> would add an<br>
> absolute time limit on ringing phone calls to 60 seconds,<br>
> which I<br>
> believe is not the actual purpose of this limit. Could anyone<br>
> please<br>
> share their thoughts on this matter?<br>
><br>
><br>
> Thanks,<br>
> Mikael<br>
><br>
><br>
><br>
><br>
><br>
> _______________________________________________<br>
> Freeswitch-users mailing list<br>
> <a href="mailto:Freeswitch-users@lists.freeswitch.org" target="_blank">Freeswitch-users@lists.freeswitch.org</a><br>
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> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
><br>
><br>
><br>
> --<br>
> Anthony Minessale II<br>
><br>
> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
> ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
><br>
> AIM: anthm<br>
> <a href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>
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><br>
> FreeSWITCH Developer Conference<br>
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<br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br>
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<a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
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