Ah, just found the culprit. Stupid bug! http://trac.pjsip.org/repos/ticket/412#comment:4 Thanks for pointing out the exact revision that contains the bug. cheers, -benny On 2/28/08, Dan <dan.aberg at keystream.se> wrote: > I mean freeze as in the GUI does not respond (instance2, the callee). > I checked earlier versions and the behaviour I experience starts with > revision 1815. Revision 1814 work well for me. One change between 1814 > and 1815 is the file "pjlib/src/pj/guid_simple.c". So when I copied this > file from revision 1814 to revision 1826 it worked like a charm. > > As for a log file, I can't use the "--log-file" parameter since I have > to kill the process for it to stop, and then the log file is empty. But > here's a cut-and-paste from the console window of the callee that > freezes: > > -- Start of dump --- > 22:06:41.387 os_core_unix.c pjlib 0.8.0-trunk for POSIX initialized > 22:06:41.389 sip_endpoint.c Creating endpoint instance... > 22:06:41.391 pjlib select() I/O Queue created (0xb7b57098) > 22:06:41.391 sip_endpoint.c Module "mod-msg-print" registered > 22:06:41.391 sip_transport. Transport manager created. > 22:06:41.392 sip_endpoint.c Module "mod-pjsua-log" registered > 22:06:41.392 sip_endpoint.c Module "mod-tsx-layer" registered > 22:06:41.392 sip_endpoint.c Module "mod-stateful-util" registered > 22:06:41.393 sip_endpoint.c Module "mod-ua" registered > 22:06:41.393 sip_endpoint.c Module "mod-100rel" registered > 22:06:41.393 sip_endpoint.c Module "mod-pjsua" registered > 22:06:41.393 sip_endpoint.c Module "mod-invite" registered > 22:06:41.614 pasound.c PortAudio sound library initialized, > status=0 > 22:06:41.615 pasound.c PortAudio host api count=2 > 22:06:41.615 pasound.c Sound device count=11 > 22:06:41.615 pjlib select() I/O Queue created (0x81c355c) > 22:06:41.616 conference.c Creating conference bridge with 254 ports > 22:06:41.616 conference.c Sound device successfully created for port > 0 > 22:06:41.616 sip_endpoint.c Module "mod-evsub" registered > 22:06:41.616 sip_endpoint.c Module "mod-presence" registered > 22:06:41.616 evsub.c Event pkg "presence" registered by > mod-presence > 22:06:41.616 sip_endpoint.c Module "mod-refer" registered > 22:06:41.616 evsub.c Event pkg "refer" registered by mod-refer > 22:06:41.616 sip_endpoint.c Module "mod-pjsua-pres" registered > 22:06:41.616 sip_endpoint.c Module "mod-pjsua-im" registered > 22:06:41.617 sip_endpoint.c Module "mod-pjsua-options" registered > 22:06:41.617 pjsua_core.c 1 SIP worker threads created > 22:06:41.617 pjsua_core.c pjsua version 0.8.0-trunk for > i686-pc-linux-gnu initialized > 22:06:41.617 pjsua_core.c SIP UDP socket reachable at > 192.168.27.101:5060 > 22:06:41.617 udp0x81cda18 SIP UDP transport started, published > address is 192.168.27.101:5060 > 22:06:41.617 pjsua_acc.c Account <sip:192.168.27.101:5060> added > with id 0 > 22:06:41.618 tcplis:5060 SIP TCP listener ready for incoming > connections at 192.168.27.101:5060 > 22:06:41.618 pjsua_acc.c Account > <sip:192.168.27.101:5060;transport=TCP> added with id 1 > 22:06:41.618 pjsua_media.c RTP socket reachable at 192.168.27.101:4000 > 22:06:41.618 pjsua_media.c RTCP socket reachable at > 192.168.27.101:4001 > 22:06:41.619 pjsua_media.c RTP socket reachable at 192.168.27.101:4002 > 22:06:41.619 pjsua_media.c RTCP socket reachable at > 192.168.27.101:4003 > 22:06:41.619 pjsua_media.c RTP socket reachable at 192.168.27.101:4004 > 22:06:41.619 pjsua_media.c RTCP socket reachable at > 192.168.27.101:4005 > 22:06:41.619 pjsua_media.c RTP socket reachable at 192.168.27.101:4006 > 22:06:41.619 pjsua_media.c RTCP socket reachable at > 192.168.27.101:4007 > 22:06:41.620 pjsua_media.c Opening null sound device.. > >>>> > Account list: > [ 0] <sip:192.168.27.101:5060>: does not register > Online status: Online > *[ 1] <sip:192.168.27.101:5060;transport=TCP>: does not register > Online status: Online > Buddy list: > -none- > > +=============================================================================+ > | 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 > >>> 22:06:52.663 sip_endpoint.c Processing incoming message: Request > msg INVITE/cseq=22862 (rdata0x81cde8c) > 22:06:52.663 pjsua_core.c RX 1013 bytes Request msg INVITE/cseq=22862 > (rdata0x81cde8c) from UDP 127.0.0.1:6028: > INVITE sip:127.0.0.1 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.27.101:6028;rport;branch=z9hG4bKPjAexwNdDGcCMN6UjoiFXwqHtsovVPTsxp > Max-Forwards: 70 > From: <sip:192.168.27.101>;tag=H8sr2agDv2dDoKBZ862a80TOAxF8kQ8B > To: sip:127.0.0.1 > Contact: <sip:192.168.27.101:6028> > Call-ID: sikukbFF7MWWeztHmRWRE0O.3dpb17Tz > CSeq: 22862 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > PUBLISH, REFER, MESSAGE, OPTIONS > Supported: replaces, 100rel, norefersub > User-Agent: PJSUA v0.8.0-trunk/i686-pc-linux-gnu > Content-Type: application/sdp > Content-Length: 441 > > v=0 > o=- 3413221612 3413221612 IN IP4 192.168.27.101 > s=pjmedia > c=IN IP4 192.168.27.101 > t=0 0 > a=X-nat:0 > m=audio 4008 RTP/AVP 103 102 104 117 3 0 8 101 > a=rtcp:4009 IN IP4 192.168.27.101 > a=rtpmap:103 speex/16000 > a=rtpmap:102 speex/8000 > a=rtpmap:104 speex/32000 > a=rtpmap:117 iLBC/8000 > a=fmtp:117 mode=30 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > --end msg-- > > ---End of dump--- > > > /Dan > > > > > On Thu, 2008-02-28 at 19:10 +0000, Benny Prijono wrote: > > Ah sorry, Olle sent me a log file with Address Incomplete error and I > > thought that was the one that you're testing with. > > > > Anyway, I tested the same configuration but with --null-audio on both > > instances (since I don't have sound card in my colinux), and as > > expected, it didn't freeze. > > > > Was it really freezing, or the SIP worker thread is not responding? If > > the whole application was freezing, that's strange, since the console > > UI is in another thread so even if the worker thread gets blocked for > > some reason the console UI should still be running. This especially > > true for callee (and instance 2 in your setup is callee), unlike > > caller where the console UI thread is responsible for creating the > > call (so if make_call() gets stucked, the console UI will get stucked > > too). > > > > Can you share any logs maybe? > > > > cheers, > > -benny > > > > On 2/28/08, Dan <dan.aberg at keystream.se> wrote: > > > Actually I'm not registering with OpenSER or any other server. I just > > > run pjsua without configuration file and the parameters in my originally > > > post. > > > Anyway the second pjsua instance freezes, I don't think it should do > > > that, strict route or not :) > > > > > > > > > /Dan > > > > > > > > > On Thu, 2008-02-28 at 15:36 +0000, Benny Prijono wrote: > > > > On 2/28/08, Dan <dan.aberg at keystream.se> wrote: > > > > > Hi, > > > > > I just checked out the latest revision(1824), and now I'm not able to > > > > > make a call between two instances running on the sam host. > > > > > > > > > > Instance 1: > > > > > ./pjsua-i686-pc-linux-gnu --local-port=6028 --no-vad --ec-tail=0 > > > > > --auto-answer=200 > > > > > > > > > > Instance 2: > > > > > ./pjsua-i686-pc-linux-gnu --auto-loop --no-vad --ec-tail=0 > > > > > --auto-answer=200 --null-audio > > > > > > > > > > When I make a call from instance 1 to sip:127.0.0.1, instance 2 freezes > > > > > after receiving the initial INVITE. > > > > > > > > > > The last time I tried this was with revision 1786, which worked just > > > > > fine. > > > > > > > > Hi Dan, > > > > > > > > >From the log (off list), it seems that the call was rejected with 484 > > > > Address Incomplete, because the request was like this: > > > > > > > > INVITE sip:domain SIP/2.0 > > > > Route: <sip:400 at domain> > > > > > > > > So you're using strict route, and it seems that your OpenSER couldn't > > > > handle this (maybe a configuration problem?). > > > > > > > > This worked in r1786, because we had this bug: > > > > http://trac.pjsip.org/repos/ticket/492, which incorrectly swapped back > > > > the Route with the request URI after the request is challenged with > > > > 401/407. > > > > > > > > So the solution is to use loose route, by adding ";lr" in your route URI. > > > > > > > > cheers, > > > > -benny > > > > > > > > > > > > > Thanks, > > > > > Dan > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > 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 > > > > > > > _______________________________________________ > > 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 >
