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
> > > >
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> > > > pjsip at lists.pjsip.org
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> > > >
> > >
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> > >
> > > pjsip mailing list
> > > pjsip at lists.pjsip.org
> > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> >
> >
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> >
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> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> >
>
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