taufiqsp webrtc2sip issues

taufiqsp / webrtc2sip

Automatically exported from code.google.com/p/webrtc2sip

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issues

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Webrtc2sip crashing randomly in SIPML5 scenario

#185 GoogleCodeExporter opened 8 years ago
0
how to make sip client direct to gateway ?

#184 GoogleCodeExporter opened 8 years ago
1
Webrtc2sip crash when media coder is enabled.

#183 GoogleCodeExporter opened 8 years ago
0
183/early media not forwarded to browser by webrtc2sip

#182 GoogleCodeExporter opened 8 years ago
0
Unable to start webrtc2sip using init.d script

#181 GoogleCodeExporter opened 8 years ago
0
asterisk 11.2.1+sipml5+webrtc2sip - No audio in chrome, audio available in firefox

#180 GoogleCodeExporter opened 8 years ago
1
Problem in BYE send from Webrtc2sip (Version 2.6.0)

#179 GoogleCodeExporter opened 8 years ago
1
Webrtc2sip Installing but net/if_dl.h / net/if_types.h not found in doubango compiling

#178 GoogleCodeExporter opened 8 years ago
0
webrtc2sip crash on call hold from chrome browser

#177 GoogleCodeExporter opened 8 years ago
0
Assertion Failure while running voice call

#176 GoogleCodeExporter opened 8 years ago
0
stun server installation for webrtc2sip

#175 GoogleCodeExporter opened 8 years ago
1
Webrtc2sip on Ubuntu 14.04 not working

#174 GoogleCodeExporter opened 8 years ago
6
webrtc2sip certificate issues

#173 GoogleCodeExporter opened 8 years ago
2
webrtc2sip + sipml5 : MSG: WS handshaking not done yet

#172 GoogleCodeExporter opened 8 years ago
0
webrtc2sip build error - ‘strnicmp’ was not declared in this scope

#171 GoogleCodeExporter closed 8 years ago
1
webrtc2sip build error - ‘strnicmp’ was not declared in this scope

#170 GoogleCodeExporter closed 8 years ago
1
crash when closing connection

#169 GoogleCodeExporter closed 8 years ago
1
webrtc2sip fails to build: undefined references

#168 GoogleCodeExporter closed 8 years ago
3
no Sound, getting errors on commandline

#167 GoogleCodeExporter opened 8 years ago
1
ACK to 401 Sent to Wrong IP

#166 GoogleCodeExporter opened 8 years ago
1
webrtc2sip: ../src/pj/os_core_unix.c:674: pj_thread_this: Assertion `!"Calling pjlib from unknown/external thread. You must " "register external threads with pj_thread_register() " "before calling any pjlib functions."' failed.

#165 GoogleCodeExporter closed 8 years ago
2
webtc2sip crash on re-INVITE

#164 GoogleCodeExporter opened 8 years ago
0
Current test for "have libs" in configure.ac (line 114) expects 13 "yes", but 14 are required

#163 GoogleCodeExporter closed 8 years ago
1
Configure --with-ffmpeg fails

#162 GoogleCodeExporter opened 8 years ago
1
exception occurs in webrtc2sip when caller hangs up.

#161 GoogleCodeExporter closed 8 years ago
1
RTCweb breaker includes invalid ice-pwd attribute in SDP

#160 GoogleCodeExporter closed 8 years ago
3
DTLS handshake failed [error:14102418:SSL routines:DTLS1_READ_BYTES:tlsv1 alert unknown ca]

#159 GoogleCodeExporter opened 8 years ago
1
SRTP establishing problem when webbreaker=true

#158 GoogleCodeExporter opened 8 years ago
0
No Ciphers Available when attempting WSS between SipML 1.3 / 1.5 and WebRTC 2.6.0

#157 GoogleCodeExporter opened 8 years ago
6
configure searches for doubango 2.0.1089, only 2.0.1086 available

#156 GoogleCodeExporter closed 8 years ago
2
webrtc2sip configure failed

#155 GoogleCodeExporter closed 8 years ago
2
webrtc2sip crash

#154 GoogleCodeExporter closed 8 years ago
1
Call not being dropped after error message received on Firefox 26

#153 GoogleCodeExporter opened 8 years ago
0
webrtc2sip is not sending 200 OK to SIP MESSAGE sender for a delivered SIP MESSAGE where both users are served by webrtc2sip gw supported by a SIP server in the back-end

#152 GoogleCodeExporter opened 8 years ago
3
Ability to configure the SIP Proxy (IP and port) from webrtc2sip server configuration file

#151 GoogleCodeExporter opened 8 years ago
0
random crashs

#150 GoogleCodeExporter closed 8 years ago
1
No Media (audio) When The Call Gets Connected to Voice Mail on a SIP Server

#149 GoogleCodeExporter opened 8 years ago
0
Disconnected/Unautorized running siplm5 live demo with sip2sip recommended

#148 GoogleCodeExporter opened 8 years ago
1
SDP Parse Error

#147 GoogleCodeExporter opened 8 years ago
0
Private extensions

#146 GoogleCodeExporter closed 8 years ago
1
MSG: Failed to start the RTP/RTCP transport

#145 GoogleCodeExporter opened 8 years ago
0
When will be a new release?

#144 GoogleCodeExporter closed 8 years ago
1
no audio when using mobile connection

#143 GoogleCodeExporter opened 8 years ago
2
Stun? or webRTC

#142 GoogleCodeExporter opened 8 years ago
0
Video V8 freez

#141 GoogleCodeExporter opened 8 years ago
0
webrtc2sip: double free or corruption (fasttop)

#140 GoogleCodeExporter closed 8 years ago
4
Connection UDP

#139 GoogleCodeExporter opened 8 years ago
0
WebRTC2SIP error in code

#138 GoogleCodeExporter closed 8 years ago
1
Using stun server leads to termination of call abruptly (WebRTC Gateway)

#137 GoogleCodeExporter opened 8 years ago
1
webrtc2sip 2.6.0 not started

#136 GoogleCodeExporter opened 8 years ago
4