|
Hi Johan:
This has been resolved. The problem was due to our FreeSwitch server sending INFO messages to the client every time it tries for a session update (audio to video and vice versa). We added some logs to see what was causing the content length to go beyond 1000 bytes, and were surprised to see the buffer being passed as an argument to calculate_new_content_length function. Here it is:
v=0
o=FreeSWITCH 1494550159 1494550161 IN IP4 XX.XX.44.166
s=FreeSWITCH
c=IN IP4 XX.XX.44.166
t=0 0
m=audio 32392 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=rtcp:32393 IN IP4 XX.XX.44.166
m=video 30442 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1
INFO sip:104@[2001:2::aab1:f863:a967:5bab:f24]:62007;transport=TCP;ob SIP/2.0
Via: SIP/2.0/TCP XX.XX.44.166;branch=z9hG4bKDFgHZDr6Kpvep
Max-Forwards: 70
From: <sip:103@xxxxxxxxxxxxxxxxxxxx>;tag=SQ8ervgXBXrXF
To: <sip:104@xxxxxxxxxxxxxxxxxxxx>;tag=cqaxfyCBeIKml95wpBVFbzZkWTQ0QuFB
Call-ID: mTLalvJyCpgsLz-AbJcwMx3JxcKuKh8m
CSeq: 106955541 INFO
Contact: <sip:103@xxxxx.44.166:5060;transport=tcp>
User-Agent: VoIP-PBX
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Type: application/media_control+xml
Content-Length: 175
<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>
48a49
> int len;
50a52,53
> char* body_end;
>
52a56,60
> body_end = strstr(body_start,"INFO sip");
> if(body_end != NULL) {
> len = body_end - body_start;
> return len;
> }
Regards,
Girish
From: pjsip <pjsip-bounces@xxxxxxxxxxxxxxx> on behalf of Girish Gopinath <gopinath_girish@xxxxxxxxxxx>
Sent: Wednesday, May 10, 2017 7:46 PM To: JOHAN LANTZ; pjsip list Subject: Re: NAT64 ios issue Hi Johan:
Thank you for the reply. Added logs below from an attempt to switch from audio call to video. The content length is never more than 1000, but in the logs it says 1253. Looks like I'm missing something here.
Thank you,
Girish
15:20:08.575 pjsua_core.c ....TX 1398 bytes Request msg INVITE/cseq=11944 (tdta0x12284d200) to TCP 2001:2:0:1baa::4225:2ca6:5060:
INVITE sip:mod_sofia@[2001:2:0:1baa::4225:2ca6]:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP XX.XX.117.54:47919;rport;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias Max-Forwards: 70 From: <sip:104@xxxxx.117.54;ob>;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB To: "SipApp100" <sip:103@xxxxxxxxxxxxxxxxxxxx>;tag=5Zv34Z5NQ22QF Contact: <sip:104@[2001:2::aab1:292f:d2ac:6e72:35f5]:58184;transport=TCP;ob> Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f CSeq: 11944 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: VoIPUA-1.0 iOS-10.3.1/arm-iPhone8,1/iOS-SDK Content-Type: application/sdp Content-Length: 668 v=0 o=- 3703398596 3703398599 IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5 s=pjmedia b=AS:352 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 0 99 9 96 c=IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5 b=TIAS:64000 a=rtcp:4001 IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:99 speex/32000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 m=video 4004 RTP/AVP 97 c=IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5 b=TIAS:256000 a=rtcp:4005 IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5 a=sendrecv a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42e01e; packetization-mode=1 --end msg-- 15:20:08.781 pj_nat64.c !.ipv6_mod_on_rx 15:20:08.781 pj_nat64.c .Incoming INVITE or 200 OK. If they contain IPv4 addresses, we need to change to ipv6 15:20:08.781 pj_nat64.c .**********Incoming INVITE or 200 with IPv4 address************* 15:20:08.781 pj_nat64.c .SIP/2.0 100 Trying Via: SIP/2.0/TCP XX.XX.117.54:47919;rport=47919;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias From: <sip:104@xxxxx.117.54;ob>;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB To: "SipApp100" <sip:103@xxxxxxxxxxxxxxxxxxxx>;tag=5Zv34Z5NQ22QF Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f CSeq: 11944 INVITE User-Agent: Fusion-PBX Content-Length: 0 15:20:08.781 pj_nat64.c .*************************************************************** 15:20:08.781 pj_nat64.c .Scanner syntax error at 15:20:08.781 pj_nat64.c .Error: Parsing of the incoming INVITE/200 OK failed at . Leave incoming buffer as is 15:20:08.781 pjsua_core.c .RX 375 bytes Response msg 100/INVITE/cseq=11944 (rdata0x122890f30) from TCP 2001:2:0:1baa::4225:2ca6:5060: SIP/2.0 100 Trying Via: SIP/2.0/TCP XX.XX.117.54:47919;rport=47919;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias From: <sip:104@xxxxx.117.54;ob>;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB To: "SipApp100" <sip:103@xxxxxxxxxxxxxxxxxxxx>;tag=5Zv34Z5NQ22QF Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f CSeq: 11944 INVITE User-Agent: Fusion-PBX Content-Length: 0 --end msg-- 15:20:08.789 pj_nat64.c .ipv6_mod_on_rx 15:20:08.789 pj_nat64.c .Incoming INVITE or 200 OK. If they contain IPv4 addresses, we need to change to ipv6 15:20:08.789 pj_nat64.c .**********Incoming INVITE or 200 with IPv4 address************* 15:20:08.789 pj_nat64.c .SIP/2.0 200 OK Via: SIP/2.0/TCP XX.XX.117.54:47919;rport=47919;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias From: <sip:104@xxxxx.117.54;ob>;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB To: "SipApp100" <sip:103@xxxxxxxxxxxxxxxxxxxx>;tag=5Zv34Z5NQ22QF Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f CSeq: 11944 INVITE Contact: <sip:mod_sofia@xxxxx.44.166:5060;transport=tcp> User-Agent: Fusion-PBX Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 358 v=0 o=FreeSWITCH 1494392300 1494392302 IN IP4 XX.XX.44.166 s=FreeSWITCH c=IN IP4 XX.XX.44.166 t=0 0 m=audio 16812 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=rtcp:16813 IN IP4 XX.XX.44.166 m=video 23982 RTP/AVP 97 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42e01e; packetization-mode=1 15:20:08.789 pj_nat64.c .*************************************************************** 15:20:08.789 pj_nat64.c .Extracted ip4 address as XX.XX.44.166 15:20:08.793 pj_nat64.c .Extracted ip4 address as XX.XX.44.166 15:20:08.794 pj_nat64.c .Extracted ip4 address as XX.XX.44.166 15:20:08.795 pj_nat64.c .Scanner syntax error at 15:20:08.795 pj_nat64.c .Current Content-Length is: 358 and new Content-Length is 1253 . 15:20:08.795 pj_nat64.c .Updated content length needs more bytes than old one, we must do expand and copy. TODO 15:20:08.795 pj_nat64.c .We have successfully parsed the INVITE/200 OK until EOF. Replace rx buffer. pjsip will now print the modified rx packet. 15:20:08.795 pj_nat64.c .Host in Contact header is XX.XX.44.166 15:20:08.796 pjsua_core.c .RX 1914 bytes Response msg 200/INVITE/cseq=11944 (rdata0x122890f30) from TCP 2001:2:0:1baa::4225:2ca6:5060: SIP/2.0 200 OK Via: SIP/2.0/TCP XX.XX.117.54:47919;rport=47919;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias From: <sip:104@xxxxx.117.54;ob>;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB To: "SipApp100" <sip:103@xxxxxxxxxxxxxxxxxxxx>;tag=5Zv34Z5NQ22QF Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f CSeq: 11944 INVITE Contact: <sip:mod_sofia@xxxxx.44.166:5060;transport=tcp> User-Agent: Fusion-PBX Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 358 v=0 o=FreeSWITCH 1494392300 1494392302 IN IP6 2001:2:0:1baa::4225:2ca6 s=FreeSWITCH c=IN IP6 2001:2:0:1baa::4225:2ca6 t=0 0 m=audio 16812 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=rtcp:16813 IN IP6 2001:2:0:1baa::4225:2ca6 m=video 23982 RTP/AVP 97 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42e01e; packetization-mode=1 --end msg-- 15:20:08.797 sdp.c ....Error parsing SDP in line 15 col 0: Success Assertion failed: (ctx.last_error != PJ_SUCCESS), function pjmedia_sdp_parse, file ../src/pjmedia/sdp.c, line 1349. (lldb) From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
Sent: Wednesday, May 10, 2017 7:00 PM To: Girish Gopinath; pjsip list Subject: Re: NAT64 ios issue Could you send an example of the sdp you are processing?
I assume this simply means that previously your content length used for instance 3 chars, something like 980 and then after the manipulation content length is more than 1000 bytes meaning we need 4 characters instead of 3. IIRC my sdp´s were quite far
away from the limit so I added that scenario as a corner case that I did not implement. I think you will be able to fix it yourself if you spend some time on it. I am going to spend a little time on this code this week but I can not promise I have time to
add this scenario so I suggest you start by trying to fix it yourself (and contribute the fix back to GitHub if it works :-)
Johan
From: Girish Gopinath
Date: Wednesday, 10 May 2017 at 15:08 To: pjsip list, Johan Lantz Subject: Re: NAT64 ios issue Este mensaje y sus adjuntos se dirigen exclusivamente a su destinatario, puede contener información privilegiada o confidencial y es para uso exclusivo de la persona o entidad de destino. Si no es usted. el destinatario indicado, queda notificado de que la lectura, utilización, divulgación y/o copia sin autorización puede estar prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it. Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição |
Attachment:
pj-nat64.c.patch
Description: pj-nat64.c.patch
_______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@xxxxxxxxxxxxxxx http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
