ChallyCai/webrtc2sip

Issues

• 183/early media not forwarded to browser by webrtc2sip

  #182 opened 9 years ago by GoogleCodeExporter
  0

• Webrtc2sip crash when media coder is enabled.

  #183 opened 9 years ago by GoogleCodeExporter
  0

• Problem in BYE send from Webrtc2sip (Version 2.6.0)

  #179 opened 9 years ago by GoogleCodeExporter
  1

• asterisk 11.2.1+sipml5+webrtc2sip - No audio in chrome, audio available in firefox

  #180 opened 9 years ago by GoogleCodeExporter
  1

• Unable to start webrtc2sip using init.d script

  #181 opened 9 years ago by GoogleCodeExporter
  0

• webrtc2sip crash on call hold from chrome browser

  #177 opened 9 years ago by GoogleCodeExporter
  0

• Webrtc2sip Installing but net/if_dl.h / net/if_types.h not found in doubango compiling

  #178 opened 9 years ago by GoogleCodeExporter
  0

• stun server installation for webrtc2sip

  #175 opened 9 years ago by GoogleCodeExporter
  1

• Assertion Failure while running voice call

  #176 opened 9 years ago by GoogleCodeExporter
  0

• Webrtc2sip on Ubuntu 14.04 not working

  #174 opened 9 years ago by GoogleCodeExporter
  4

• webrtc2sip + sipml5 : MSG: WS handshaking not done yet

  #172 opened 9 years ago by GoogleCodeExporter
  0

• webrtc2sip certificate issues

  #173 opened 9 years ago by GoogleCodeExporter
  2

• webrtc2sip build error - ‘strnicmp’ was not declared in this scope

  #170 opened 9 years ago by GoogleCodeExporter
  1

• webrtc2sip build error - ‘strnicmp’ was not declared in this scope

  #171 opened 9 years ago by GoogleCodeExporter
  1

• crash when closing connection

  #169 opened 9 years ago by GoogleCodeExporter
  1

• webrtc2sip fails to build: undefined references

  #168 opened 9 years ago by GoogleCodeExporter
  3

• ACK to 401 Sent to Wrong IP

  #166 opened 9 years ago by GoogleCodeExporter
  1

• no Sound, getting errors on commandline

  #167 opened 9 years ago by GoogleCodeExporter
  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 opened 9 years ago by GoogleCodeExporter
  2

• Current test for "have libs" in configure.ac (line 114) expects 13 "yes", but 14 are required

  #163 opened 9 years ago by GoogleCodeExporter
  1

• webtc2sip crash on re-INVITE

  #164 opened 9 years ago by GoogleCodeExporter
  0

• Configure --with-ffmpeg fails

  #162 opened 9 years ago by GoogleCodeExporter
  1

• exception occurs in webrtc2sip when caller hangs up.

  #161 opened 9 years ago by GoogleCodeExporter
  1

• RTCweb breaker includes invalid ice-pwd attribute in SDP

  #160 opened 9 years ago by GoogleCodeExporter
  3

• DTLS handshake failed [error:14102418:SSL routines:DTLS1_READ_BYTES:tlsv1 alert unknown ca]

  #159 opened 9 years ago by GoogleCodeExporter
  1

• No Ciphers Available when attempting WSS between SipML 1.3 / 1.5 and WebRTC 2.6.0

  #157 opened 9 years ago by GoogleCodeExporter
  6

• SRTP establishing problem when webbreaker=true

  #158 opened 9 years ago by GoogleCodeExporter
  0

• configure searches for doubango 2.0.1089, only 2.0.1086 available

  #156 opened 9 years ago by GoogleCodeExporter
  2

• webrtc2sip crash

  #154 opened 9 years ago by GoogleCodeExporter
  1

• webrtc2sip configure failed

  #155 opened 9 years ago by GoogleCodeExporter
  2

• Call not being dropped after error message received on Firefox 26

  #153 opened 9 years ago by GoogleCodeExporter
  0

• Ability to configure the SIP Proxy (IP and port) from webrtc2sip server configuration file

  #151 opened 9 years ago by GoogleCodeExporter
  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 opened 9 years ago by GoogleCodeExporter
  3

• random crashs

  #150 opened 9 years ago by GoogleCodeExporter
  1

• Disconnected/Unautorized running siplm5 live demo with sip2sip recommended

  #148 opened 9 years ago by GoogleCodeExporter
  1

• No Media (audio) When The Call Gets Connected to Voice Mail on a SIP Server

  #149 opened 9 years ago by GoogleCodeExporter
  0

• Private extensions

  #146 opened 9 years ago by GoogleCodeExporter
  1

• SDP Parse Error

  #147 opened 9 years ago by GoogleCodeExporter
  0

• no audio when using mobile connection

  #143 opened 9 years ago by GoogleCodeExporter
  2

• When will be a new release?

  #144 opened 9 years ago by GoogleCodeExporter
  1

• MSG: Failed to start the RTP/RTCP transport

  #145 opened 9 years ago by GoogleCodeExporter
  0

• Stun? or webRTC

  #142 opened 9 years ago by GoogleCodeExporter
  0

• webrtc2sip: double free or corruption (fasttop)

  #140 opened 9 years ago by GoogleCodeExporter
  4

• Video V8 freez

  #141 opened 9 years ago by GoogleCodeExporter
  0

• Connection UDP

  #139 opened 9 years ago by GoogleCodeExporter
  0

• WebRTC2SIP error in code

  #138 opened 9 years ago by GoogleCodeExporter
  1

• Using stun server leads to termination of call abruptly (WebRTC Gateway)

  #137 opened 9 years ago by GoogleCodeExporter
  1

• duplicate symbol when building in MAC OSX 10.8.5

  #135 opened 9 years ago by GoogleCodeExporter
  1

• webrtc2sip 2.6.0 not started

  #136 opened 9 years ago by GoogleCodeExporter
  4

• doubango and webrtc2sip configure.ac mismatch in svn r118 causes build to fail

  #134 opened 9 years ago by GoogleCodeExporter
  1