JiaNemo / idoubs

Automatically exported from code.google.com/p/idoubs
0 stars 0 forks source link

Outgoing calls with tcp #187

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago
a) Before posting your issue you MUST answer to the questions otherwise it
will be rejected (invalid status) by us
b) Please check the issue tacker to avoid duplication
c) Please provide network capture (wireshark) or Android log (DDMS output)
if you want quick response

Outgoing calls not working with tcp. Incoming calls are ok.

What steps will reproduce the problem?
1. set transport to tcp
2. run the app
3. try to call

What is the expected output? What do you see instead?

Expected to call remote number, instead call hangs up and softphone disconnects 
from server

What version of the product or source code revision are you using? On what
operating system?

latest idoubs version

Please provide any additional information below.

*INFO: State machine: c0000_Started_2_Outgoing_X_oINVITE
*INFO: tdav_consumer_audio_init()
*INFO: Create SpeexDSP jitter buffer
*INFO: 'srtp-mode' param ignored beacuse SRTP not enabled. Please rebuild the 
source code with this option.
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: 

SEND: INVITE sip:armatbk@sip.linphone.org SIP/2.0
Via: SIP/2.0/TCP 172.20.3.34:56447;branch=z9hG4bK-679100694;rport
From: <sip:aterite@sip.linphone.org>;tag=2137951623
To: <sip:armatbk@sip.linphone.org>
Contact: 
<sip:aterite@172.20.3.34:56447;transport=tcp>;+g.oma.sip-im;language="en,fr";+g.
3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: a68bebaa-3054-ebac-cba8-324317101257
CSeq: 1038479011 INVITE
Content-Type: application/sdp
Content-Length: 640
Max-Forwards: 70
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 ios-ngn-stack/v00 (doubango r000)
P-Preferred-Identity: <sip:aterite@sip.linphone.org>
Supported: 100rel

v=0
o=doubango 1983 678901 IN IP4 172.20.3.34
s=-
c=IN IP4 172.20.3.34
t=0 0
m=audio 36270 RTP/AVP 8 0 3 111 101
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000; sprop-maxcapturerate=16000; stereo=0; 
sprop-stereo=0; useinbandfec=0; usedtx=0
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux
a=ssrc:1273631426 cname:56fa104295d915c9a9e080ac035da107
a=ssrc:1273631426 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1273631426 label:doubango@audio

*INFO: Add call-id = 'a68bebaa-3054-ebac-cba8-324317101257' to peer with local 
fd = 18
*INFO: __CFWriteStreamClientCallBack --> kCFStreamEventCanAcceptBytes(fd=18)
*INFO: OnDialogEvent(Dialog connecting, 2)
*INFO: State machine: x9998_Any_2_Terminated_X_transportError
*INFO: === INVITE Dialog terminated ===
**WARN: function: "tsk_fsm_act()" 
file: 
"/Users/aterite/Documents/projects/iOS/newDoubango/doubango/branches/2.0/doubang
o/tinySAK/src/tsk_fsm.c" 
line: "133" 
MSG: The FSM is in the final state.
*INFO: === ICT terminated ===
*INFO: OnDialogEvent(Transport error, 2)
*INFO: [Stream] Removed call-id = 'a68bebaa-3054-ebac-cba8-324317101257' from 
peer with local fd = 18
*INFO: [Transport] Removed call-id = 'a68bebaa-3054-ebac-cba8-324317101257' 
from transport with type = 9
*INFO: [Transport Layer] Removed call-id = 
'a68bebaa-3054-ebac-cba8-324317101257' from transport layer
*INFO: *** tdav_session_audio_t destroyed ***
*INFO: Not started
*INFO: CloseSocket(22)
*INFO: CloseSocket(23)
*INFO: Not started
*INFO: *** SpeexDSP denoiser destroyed ***
*INFO: *** SpeexDSP jb destroyed ***
*INFO: *** AudioUnit Consumer destroyed ***
*INFO: *** AudioUnit Producer destroyed ***
*INFO: *** RTP manager destroyed ***
*INFO: *** Audio session destroyed ***
*INFO: *** INVITE Dialog destroyed ***
*INFO: *** ICT destroyed ***
*INFO: *** SIP Session destroyed ***

Original issue reported on code.google.com by arma...@gmail.com on 5 Nov 2014 at 12:22

GoogleCodeExporter commented 9 years ago

Original comment by boss...@yahoo.fr on 6 Nov 2014 at 2:37

GoogleCodeExporter commented 9 years ago
it's not a duplicate, it's another problem with tcp

Original comment by arma...@gmail.com on 6 Nov 2014 at 4:30

GoogleCodeExporter commented 9 years ago
I thing this issue somehow related to this
https://code.google.com/p/doubango/issues/detail?id=145

Original comment by arma...@gmail.com on 6 Nov 2014 at 5:54

GoogleCodeExporter commented 9 years ago
Also this https://code.google.com/p/imsdroid/issues/detail?id=417

Original comment by arma...@gmail.com on 6 Nov 2014 at 5:55