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?
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...
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