On 1/15/08, Alric Silvera <alric.silvera at yahoo.com> wrote: > > Hello > > To better understand PJSIP, i was going through the samples after a > successful standard build. I liked the sipstateless sample. But I am unable > to get it to work. I have been using the same machine, but using different > ports. The terminal trace for each app is listed below. The sipstateless app > just does not respond. Nor, do you see the 501's on pjsua. Am I overlooking > something? > Everything looks fine from the log, so I'm not sure why the request didn't reach sipstateless. What if you make call to sip:localhost instead? cheers, -benny Thanks > > > Running PJSIP - PJSUA.... > > alric at debian:~/Code/pjproject-0.8.0/pjsip-apps/bin$ > ./pjsua-i686-pc-linux-gnu --local-port=5062 > 23:45:07.162 os_core_unix.c pjlib 0.8.0 for POSIX initialized > 23:45:07.163 sip_endpoint.c Creating endpoint instance... > 23:45:07.163 pjlib select() I/O Queue created (0x81596ac) > 23:45:07.163 sip_endpoint.c Module "mod-msg-print" registered > 23:45:07.163 sip_transport. Transport manager created. > 23:45:07.163 sip_endpoint.c Module "mod-pjsua-log" registered > 23:45:07.163 sip_endpoint.c Module "mod-tsx-layer" registered > 23:45:07.163 sip_endpoint.c Module "mod-stateful-util" registered > 23:45:07.163 sip_endpoint.c Module "mod-ua" registered > 23:45:07.163 sip_endpoint.c Module "mod-100rel" registered > 23:45:07.164 sip_endpoint.c Module "mod-pjsua" registered > 23:45:07.164 sip_endpoint.c Module "mod-invite" registered > 23:45:07.168 pasound.c PortAudio sound library initialized, status=0 > 23:45:07.168 pasound.c PortAudio host api count=1 > 23:45:07.168 pasound.c Sound device count=1 > 23:45:07.168 pjlib select() I/O Queue created (0x819237c) > 23:45:07.169 sip_endpoint.c Module "mod-evsub" registered > 23:45:07.169 sip_endpoint.c Module "mod-presence" registered > 23:45:07.169 sip_endpoint.c Module "mod-refer" registered > 23:45:07.169 sip_endpoint.c Module "mod-pjsua-pres" registered > 23:45:07.169 sip_endpoint.c Module "mod-pjsua-im" registered > 23:45:07.169 sip_endpoint.c Module "mod-pjsua-options" registered > 23:45:07.169 pjsua_core.c 1 SIP worker threads created > 23:45:07.169 pjsua_core.c pjsua version 0.8.0 for i686-pc-linux-gnu > initialized > 23:45:07.170 pjsua_core.c SIP UDP socket reachable at > 192.168.0.102:5062 > 23:45:07.170 udp0x81aea34 SIP UDP transport started, published address > is 192.168.0.102:5062 > 23:45:07.170 pjsua_acc.c Account <sip:192.168.0.102:5062;transport=UDP> > added with id 0 > 23:45:07.170 tcplis:5062 SIP TCP listener ready for incoming > connections at 192.168.0.102:5062 > 23:45:07.170 pjsua_acc.c Account <sip:192.168.0.102:5062;transport=TCP> > added with id 1 > 23:45:07.170 pjsua_media.c RTP socket reachable at 192.168.0.102:4000 > 23:45:07.170 pjsua_media.c RTCP socket reachable at 192.168.0.102:4001 > 23:45:07.170 pjsua_media.c RTP socket reachable at 192.168.0.102:4002 > 23:45:07.170 pjsua_media.c RTCP socket reachable at 192.168.0.102:4003 > 23:45:07.170 pjsua_media.c RTP socket reachable at 192.168.0.102:4004 > 23:45:07.170 pjsua_media.c RTCP socket reachable at 192.168.0.102:4005 > 23:45:07.170 pjsua_media.c RTP socket reachable at 192.168.0.102:4006 > 23:45:07.170 pjsua_media.c RTCP socket reachable at 192.168.0.102:4007 > 23:45:07.170 pjsua_media.c pjsua_set_snd_dev(): attempting to open > devices @16000 Hz > 23:45:07.250 os_core_unix.c Info: possibly re-registering existing thread > 23:45:07.352 echo_speex.c Speex Echo canceller/AEC created, > clock_rate=16000, samples per frame=160, tail length=200 ms, latency=32 ms > >>>> > Account list: > [ 0] <sip:192.168.0.102:5062;transport=UDP>: does not register > Online status: Online > *[ 1] <sip:192.168.0.102:5062;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 > >>> 23:45:12.416 sound_port.c EC suspended because of inactivity > > >>>> > Account list: > [ 0] <sip:192.168.0.102:5062;transport=UDP>: does not register > Online status: Online > *[ 1] <sip:192.168.0.102:5062;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 > >>> m > (You currently have 0 calls) > Buddy list: > -none- > > Choices: > 0 For current dialog. > -1 All 0 buddies in buddy list > [1 - 0] Select from buddy list > URL An URL > <Enter> Empty input (or 'q') to cancel > Make call: sip:192.168.0.102 > 23:45:36.050 pjsua_call.c Making call with acc #1 to sip:192.168.0.102 > 23:45:36.050 pjsua_core.c TX 952 bytes Request msg INVITE/cseq=6249 > (tdta0x81e0fa0) to UDP 192.168.0.102:5060: > INVITE sip:192.168.0.102 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.102:5062 > ;rport;branch=z9hG4bKPj0fd40000000367458b6b > Max-Forwards: 70 > From: <sip:192.168.0.102>;tag=0fd40000000167458b6b > To: sip:192.168.0.102 > Contact: <sip:192.168.0.102:5062;transport=UDP> > Call-ID: 0fd40000000267458b6b > CSeq: 6249 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > PUBLISH, REFER, MESSAGE, OPTIONS > Supported: replaces, 100rel, norefersub > User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu > Content-Type: application/sdp > Content-Length: 404 > > v=0 > o=- 3409278336 3409278336 IN IP4 192.168.0.102 > s=pjmedia > c=IN IP4 192.168.0.102 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 101 > a=rtpmap:103 speex/16000 > a=rtpmap:102 speex/8000 > a=rtpmap:104 speex/32000 > a=rtpmap:117 iLBC/8000 > a=fmtp:117 mode=20 > 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-- > . > > ...retries snipped... > . > > 23:46:08.053 pjsua_app.c Call 0 is DISCONNECTED [reason=408 (Request > Timeout)] > > > Running PJSIP Sample - sipstateless... > > alric at debian:~/Code/pjproject-0.8.0/pjsip-apps/bin/samples$ > ./sipstateless-i686-pc-linux-gnu > 23:42:59.081 os_core_unix.c pjlib 0.8.0 for POSIX initialized > 23:42:59.081 sip_endpoint.c Creating endpoint instance... > 23:42:59.081 pjlib select() I/O Queue created (0x807c6a4) > 23:42:59.081 sip_endpoint.c Module "mod-msg-print" registered > 23:42:59.081 sip_transport. Transport manager created. > 23:42:59.082 udp0x8090b4c SIP UDP transport started, published address > is 192.168.0.102:5060 > 23:42:59.082 tcplis:5060 SIP TCP listener ready for incoming > connections at 192.168.0.102:5060 > 23:42:59.082 sip_endpoint.c Module "mod-app" registered > 23:42:59.082 sipstateless.c Press Ctrl-C to quit.. > > End of Message... > > ------------------------------ > Never miss a thing. Make Yahoo your homepage.<http://us.rd.yahoo.com/evt=51438/*http://www.yahoo.com/r/hs> > > > ------------------------------ > Never miss a thing. Make Yahoo your homepage.<http://us.rd.yahoo.com/evt=51438/*http://www.yahoo.com/r/hs> > > > ------------------------------ > Never miss a thing. Make Yahoo your homepage.<http://us.rd.yahoo.com/evt=51438/*http://www.yahoo.com/r/hs> > > _______________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080116/c1e39c18/attachment-0001.html
