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...
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@xxxxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxxxx>;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
>
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