On 3/19/08, Sasa Coh <sasacoh at gmail.com> wrote: > Hello! > > Yes, it should be error in UAS. Thanks Klaus! > > Now I have another case: > While testing pjSIP's behavior using TCP transport with pjsua, I've > encountered another strange situation. If I use STUN server and UDP > transport, pjsua sends (as I expected) message SIP REGISTER with correct, > STUN resolved addresses in fields "via" and "contact". > On the other hand, if I switch to TCP transport (and use the same STUN > server) than pjsua sends message SIP REGISTER with "wrong - IMO" IP > addresses (my local-private) in "via" and "contact" fields and not like I > would expect with STUN resolved addresses. See log below... Yep that's intentional. STUN, as it stands for, applies only to UDP transports. For TCP/TLS transport, the only way to get around NAT is to use SIP outbound draft, which I plan to implement in the next few weeks. cheers, -benny > Thanks again, > Sasa > > > 14:54:52.332 os_core_win32. pjlib 0.8.0-trunk for win32 initialized > 14:54:52.348 sip_endpoint.c Creating endpoint instance... > 14:54:52.348 pjlib select() I/O Queue created (00D20858) > 14:54:52.348 sip_endpoint.c Module "mod-msg-print" registered > 14:54:52.348 sip_transport. Transport manager created. > 14:54:52.348 sip_endpoint.c Module "mod-pjsua-log" registered > 14:54:52.348 pjsua_core.c Nameserver 192.168.133.254 added > 14:54:52.348 sip_endpoint.c Module "mod-tsx-layer" registered > 14:54:52.348 sip_endpoint.c Module "mod-stateful-util" registered > 14:54:52.348 sip_endpoint.c Module "mod-ua" registered > 14:54:52.348 sip_endpoint.c Module "mod-100rel" registered > 14:54:52.348 sip_endpoint.c Module "mod-pjsua" registered > 14:54:52.348 sip_endpoint.c Module "mod-invite" registered > 14:54:52.348 pjsua_core.c STUN server stun.voipbuster.com resolved, > address is 80.239.235.209:3478 > 14:54:52.551 pasound.c PortAudio sound library initialized, status=0 > 14:54:52.551 pasound.c PortAudio host api count=3 > 14:54:52.551 pasound.c Sound device count=8 > 14:54:52.551 pjlib select() I/O Queue created (00C4E544) > 14:54:52.551 sip_endpoint.c Module "mod-evsub" registered > 14:54:52.551 sip_endpoint.c Module "mod-presence" registered > 14:54:52.551 sip_endpoint.c Module "mod-refer" registered > 14:54:52.551 sip_endpoint.c Module "mod-pjsua-pres" registered > 14:54:52.551 sip_endpoint.c Module "mod-pjsua-im" registered > 14:54:52.551 sip_endpoint.c Module "mod-pjsua-options" registered > 14:54:52.551 pjsua_core.c 1 SIP worker threads created > 14:54:52.551 pjsua_core.c pjsua version 0.8.0-trunk for win32 initialized > 14:54:52.707 pjsua_core.c SIP UDP socket reachable at 193.77.81.126:5060 > 14:54:52.707 udp00D4E000 SIP UDP transport started, published address is > 193.77.81.126:5060 > 14:54:52.707 pjsua_acc.c Account <sip:193.77.81.126:5060> added with id > 0 > 14:54:52.707 tcplis:5060 SIP TCP listener ready for incoming connections > at 192.168.133.10:5060 > 14:54:52.707 pjsua_acc.c Account > <sip:192.168.133.10:5060;transport=TCP> added with id 1 > 14:54:52.707 pjsua_acc.c Account > sip:1551 at int-tcp.pingtel.com;transport=tcp added with id 2 > 14:54:52.707 resolver.c Transmitting 47 bytes to NS 0 > (192.168.133.254:53): DNS SRV query for _sip._tcp.int-tcp.pingtel.com > 14:54:52.707 pjsua_acc.c Registration sent > 14:54:52.723 resolver.c Transmitting 37 bytes to NS 0 > (192.168.133.254:53): DNS A query for interop.pingtel.com > 14:54:52.739 tcpc00D56024 TCP client transport created > 14:54:52.739 tcpc00D56024 TCP transport 192.168.133.10:1813 is > connecting to 155.212.198.118:5060... > 14:54:52.739 pjsua_core.c TX 472 bytes Request msg REGISTER/cseq=9189 > (tdta00D53760) to tcp 155.212.198.118:5060: > REGISTER sip:int-tcp.pingtel.com;transport=tcp SIP/2.0 > Via: SIP/2.0/tcp > 192.168.133.10:1813;rport;branch=z9hG4bKPj363d10babf4c47ed907afe4a1d2ab443 > Max-Forwards: 70 > From: > <sip:1551 at int-tcp.pingtel.com>;tag=86a64bd2d20447c48f55020ba2cc41e8 > To: <sip:1551 at int-tcp.pingtel.com> > Call-ID: 8e926480539e4a6cade1065e41dc99bd > CSeq: 9189 REGISTER > User-Agent: PJSUA v0.8.0-trunk/win32 > Contact: <sip:1551 at 192.168.133.10:5060;transport=TCP> > Expires: 3600 > Content-Length: 0 > > > --end msg-- > 14:54:52.754 pjsua_media.c RTP socket reachable at 193.77.81.126:4000 > 14:54:52.754 pjsua_media.c RTCP socket reachable at 193.77.81.126:4001 > 14:54:52.801 pjsua_media.c RTP socket reachable at 193.77.81.126:4002 > 14:54:52.801 pjsua_media.c RTCP socket reachable at 193.77.81.126:4003 > 14:54:52.942 tcpc00D56024 TCP transport 192.168.133.10:1813 is connected > to 155.212.198.118:5060 > 14:54:53.020 pjsua_media.c RTP socket reachable at 193.77.81.126:4004 > 14:54:53.020 pjsua_media.c RTCP socket reachable at 193.77.81.126:4005 > 14:54:53.192 pjsua_core.c RX 675 bytes Response msg > 401/REGISTER/cseq=9189 (rdata00D561BC) from tcp 155.212.198.118:5060: > SIP/2.0 401 Unauthorized > From: > <sip:1551 at int-tcp.pingtel.com>;tag=86a64bd2d20447c48f55020ba2cc41e8 > To: <sip:1551 at int-tcp.pingtel.com> > Call-Id: 8e926480539e4a6cade1065e41dc99bd > Cseq: 9189 REGISTER > Via: SIP/2.0/tcp > 192.168.133.10:1813;rport=1813;branch=z9hG4bKPj363d10babf4c47ed907afe4a1d2ab443;received=193.77.81.126 > Www-Authenticate: Digest realm="interop.pingtel.com", > nonce="9b4f6d370adbbe7888264e0de9b6687c47da83ad" > User-Agent: sipX/3.9.7 sipX/registry (Linux) > Date: Fri, 14 Mar 2008 13:54:53 GMT > Contact: <sip:155.212.198.118:5070> > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE > Accept-Language: en > Supported: gruu > Content-Length: 0 > > > --end msg-- > 14:54:53.192 pjsua_core.c TX 680 bytes Request msg REGISTER/cseq=9190 > (tdta00D53760) to tcp 155.212.198.118:5060: > REGISTER sip:int-tcp.pingtel.com;transport=tcp SIP/2.0 > Via: SIP/2.0/tcp > 192.168.133.10:1813;rport;branch=z9hG4bKPjd25d246b3aba4d47a74d04e339cfbb6f > Max-Forwards: 70 > From: > <sip:1551 at int-tcp.pingtel.com>;tag=86a64bd2d20447c48f55020ba2cc41e8 > To: <sip:1551 at int-tcp.pingtel.com> > Call-ID: 8e926480539e4a6cade1065e41dc99bd > CSeq: 9190 REGISTER > User-Agent: PJSUA v0.8.0-trunk/win32 > Contact: <sip:1551 at 192.168.133.10:5060;transport=TCP> > Expires: 3600 > Authorization: Digest username="1551", realm="interop.pingtel.com", > nonce="9b4f6d370adbbe7888264e0de9b6687c47da83ad", > uri="sip:int-tcp.pingtel.com;transport=tcp", response="456ff714455d37295 > 087c37dd05f7fc6" > Content-Length: 0 > > > --end msg-- > 14:54:53.239 pjsua_media.c RTP socket reachable at 193.77.81.126:4006 > 14:54:53.239 pjsua_media.c RTCP socket reachable at 193.77.81.126:4007 > 14:54:53.426 pjsua_media.c RTP socket reachable at 193.77.81.126:4008 > 14:54:53.426 pjsua_media.c RTCP socket reachable at 193.77.81.126:4009 > 14:54:53.442 pjsua_core.c RX 606 bytes Response msg > 200/REGISTER/cseq=9190 (rdata00D561BC) from tcp 155.212.198.118:5060: > SIP/2.0 200 OK > From: > <sip:1551 at int-tcp.pingtel.com>;tag=86a64bd2d20447c48f55020ba2cc41e8 > To: <sip:1551 at int-tcp.pingtel.com>;tag=404665879 > Call-Id: 8e926480539e4a6cade1065e41dc99bd > Cseq: 9190 REGISTER > Via: SIP/2.0/tcp > 192.168.133.10:1813;rport=1813;branch=z9hG4bKPjd25d246b3aba4d47a74d04e339cfbb6f;received=193.77.81.126 > Contact: > <sip:1551 at 192.168.133.10:5060;transport=TCP>;EXPIRES=1210 > User-Agent: sipX/3.9.7 sipX/registry (Linux) > Date: Fri, 14 Mar 2008 13:54:53 GMT > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE > Accept-Language: en > Supported: gruu > Content-Length: 0 > > > --end msg-- > 14:54:53.489 pjsua_media.c RTP socket reachable at 193.77.81.126:4010 > 14:54:53.489 pjsua_media.c RTCP socket reachable at 193.77.81.126:4011 > 14:54:53.723 pjsua_media.c RTP socket reachable at 193.77.81.126:4012 > 14:54:53.723 pjsua_media.c RTCP socket reachable at 193.77.81.126:4013 > 14:54:54.020 pjsua_media.c RTP socket reachable at 193.77.81.126:4014 > 14:54:54.020 pjsua_media.c RTCP socket reachable at 193.77.81.126:4015 > 14:54:54.192 pjsua_media.c RTP socket reachable at 193.77.81.126:4016 > 14:54:54.192 pjsua_media.c RTCP socket reachable at 193.77.81.126:4017 > 14:54:54.348 pjsua_media.c RTP socket reachable at 193.77.81.126:4018 > 14:54:54.348 pjsua_media.c RTCP socket reachable at 193.77.81.126:4019 > 14:54:54.348 pjsua_media.c pjsua_set_snd_dev(): attempting to open > devices @16000 Hz > 14:54:54.348 pjsua_acc.c sip:1551 at int-tcp.pingtel.com;transport=tcp: > registration success, status=200 (OK), will re-register in 1210 seconds > >>>> > Account list: > [ 0] <sip:193.77.81.126:5060>: does not register > Online status: Online > [ 1] <sip:192.168.133.10:5060;transport=TCP>: does not register > Online status: Online > *[ 2] sip:1551 at int-tcp.pingtel.com;transport=tcp: 200/OK (expires=1204) > Online status: Online > Buddy list: > [ 1] <?> sip:1552 at int-tcp.pingtel.com;transport=tcp > [ 2] <?> sip:1553 at int-tcp.pingtel.com > > +=============================================================================+ > | Call Commands: | Buddy, IM & Presence: | Account: > | > | | | > | > | m Make new call | +b Add new buddy .| +a Add new > accnt | > | M Make multiple calls | -b Delete buddy | -a Delete > accnt. | > | a Answer call | i Send IM | !a Modify > accnt. | > | h Hangup call (ha=all) | s Subscribe presence | rr > (Re-)register | > | H Hold call | u Unsubscribe presence | ru Unregister > | > | v re-inVite (release hold) | t ToGgle Online status | > Cycle next > ac.| > | U send UPDATE | T Set online status | < Cycle prev > ac.| > | ],[ Select next/prev call > +--------------------------+-------------------+ > | x Xfer call | Media Commands: | Status & > Config: | > | X Xfer with Replaces | | > | > | # Send RFC 2833 DTMF | cl List ports | d Dump status > | > | * Send DTMF with INFO | cc Connect port | dd Dump > detailed | > | dq Dump curr. call quality | cd Disconnect port | dc Dump config > | > | | V Adjust audio Volume | f Save config > | > | S Send arbitrary REQUEST | Cp Codec priorities | f Save config > | > +------------------------------+--------------------------+-------------------+ > | q QUIT sleep N: console sleep for N ms n: detect NAT type > | > +=============================================================================+ > You have 0 active call > >>> 14:54:59.379 sound_port.c EC suspended because of inactivity > 14:55:00.770 stuntsx00D4F66 STUN timeout waiting for response > 14:55:00.770 stuntsx00D4EE0 STUN timeout waiting for response > 14:55:00.801 pjsua_app.c NAT detected as Port Restricted > > > > > On Fri, Mar 14, 2008 at 2:31 PM, Klaus Darilion > <klaus.mailinglists at pernau.at> wrote: > > > SIP/2.0 200 OK > > Via:SIP/2.0/tcp > > > 192.168.60.130:2378;branch=z9hG4bKPjbaf223aecad0451ab6f5b80e26e5143f;rport > > > From:<sip:22222239 at 192.168.104.2>;tag=1e68f3259c4242d6ba74c69ace4f5f44 > > To:<sip:2230 at 192.168.104.2>;tag=1482606729-1205334993760 > > Call-ID:5dbde9f0111241c0b3e109cce578f6a7 > > CSeq:28477 INVITE > > > Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY > > Supported:timer > > > Accept:multipart/mixed,application/media_control+xml,application/sdp > > Contact:<sip:192.168.104.2:5060> > > > > Take a look at the contact header - there is no transport parameter, > > thus the UAS announces UDP as transport for indialog requests. > > > > Either a bug or config error in the UAS. > > Or if there is a proxy in the middle which does protocol change then it > > should do record routing. > > > > regards > > klaus > > > > Sasa Coh schrieb: > > > > > Hello! > > > > > > See attachment with the pjsua log output. Btw. it works on openser but > > > not on broadworks. > > > > > > Thanks, > > > Sasa > > > > > > > > > On Wed, Mar 12, 2008 at 9:29 PM, Klaus Darilion > > > > > > > > > <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> > wrote: > > > > > > > > > > > > Sasa Coh wrote: > > > > Hello Benny, > > > > > > > > I'm testing pjSIP's behavior using pjsua (latest version) and > > > found out, > > > > that (by my opinion) pjsua has an error when using TCP transport. > > > > I was following instructions from documentation (as below): > > > > > > > > ------------------------------------------------------------------------ > > > > Using TCP Transport > > > > > > > > By default, TCP transport will be created and initialized. > > > However, TCP > > > > will not be used automatically unless the destination URL has > > > > ";transport=tcp" parameter in it. (Note: this behavior may change > > > once > > > > we support resolving NAPTR records). > > > > > > > > TCP can be specified when registering to server and when sending > > > > outgoing requests. To use TCP when registering, add > > > ";transport=tcp" in > > > > the registrar's URL, for example with "--registrar > > > sip:example.com <http://example.com> > > > > <http://example.com>;transport=tcp" option. > > > > > > > > Similarly ";transport=tcp" parameter needs to be added in the > > > > destination URL when making outgoing calls, subscribing presence, > or > > > > sending outgoing MESSAGE request. > > > > > > > > ------------------------------------------------------------------------ > > > > > > > > And now, back to the problem: > > > > - First, register via TCP - that went OK. > > > > - Then, in case of an outgoing call, everything is OK too. > > > > - The error occurs at the time when I put this call on hold. The > > > > re-INVITE (sent by pjSIP) that goes out (signaling HOLD > condition) is > > > > sent via UDP and not TCP as I would expect! > > > > > > > > Am I doing something wrong or is this an error in pjSIP? How can > I > > > > overcome this? > > > > > > Show the SIP packets (e.g. using "ngrep -W byline port 5060"). I > guess > > > there is the transport parameter missing in the contact header or in > the > > > record-route header. > > > > > > regards > > > klaus > > > > > > > > > > > Kind regards, > > > > Sasa > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > > > _______________________________________________ > > > > Visit our blog: http://blog.pjsip.org > > > > > > > > pjsip mailing list > > > > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > > > > > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > _______________________________________________ > > > Visit our blog: http://blog.pjsip.org > > > > > > pjsip mailing list > > > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > > > > > > > > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Visit our blog: http://blog.pjsip.org > > > > > > pjsip mailing list > > > pjsip at lists.pjsip.org > > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip at lists.pjsip.org > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >
