DoubangoTelecom/webrtc2sip
Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network
C
Issues
- 1
Integration with freepbx or fusionpbx
#197 opened by ugintl - 1
How to compile webrtc2sip correctly?
#194 opened by XDhughie - 0
libvpx.so.1 not found
#199 opened by llKoull - 0
Redirect call from webapp to sip phone with 302 Moved Temporarily message
#195 opened by wieczorekmichal1 - 1
- 1
Got a problem with calls
#193 opened by praeitor - 0
- 0
High CPU during calls also with same codecs
#191 opened by epasqualotto - 1
Using WSS with WebRTC and Letsencrypt (Let's Encrypt) via sipML = "no shared cipher"
#190 opened by chrischarles2002 - 1
- 2
No Audio if call is accepted after 30 seconds
#187 opened by vinodpandey23 - 0
Installed webrtc2sip but can't use the command
#189 opened by Ganasci - 0
- 4
The git directory miss the tinywrap folder and it's contents and can therefore not compile
#186 opened by legobit - 0
- 0
- 1
how to make sip client direct to gateway ?
#184 opened by GoogleCodeExporter - 0
Webrtc2sip Installing but net/if_dl.h / net/if_types.h not found in doubango compiling
#178 opened by GoogleCodeExporter - 1
asterisk 11.2.1+sipml5+webrtc2sip - No audio in chrome, audio available in firefox
#180 opened by GoogleCodeExporter - 0
- 6
Webrtc2sip on Ubuntu 14.04 not working
#174 opened by GoogleCodeExporter - 1
stun server installation for webrtc2sip
#175 opened by GoogleCodeExporter - 0
Assertion Failure while running voice call
#176 opened by GoogleCodeExporter - 0
- 2
webrtc2sip certificate issues
#173 opened by GoogleCodeExporter - 1
crash when closing connection
#169 opened by GoogleCodeExporter - 1
- 1
- 0
- 3
- 2
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 by GoogleCodeExporter - 1
ACK to 401 Sent to Wrong IP
#166 opened by GoogleCodeExporter - 1
no Sound, getting errors on commandline
#167 opened by GoogleCodeExporter - 1
Current test for "have libs" in configure.ac (line 114) expects 13 "yes", but 14 are required
#163 opened by GoogleCodeExporter - 0
webtc2sip crash on re-INVITE
#164 opened by GoogleCodeExporter - 3
- 1
- 1
Configure --with-ffmpeg fails
#162 opened by GoogleCodeExporter - 1
DTLS handshake failed [error:14102418:SSL routines:DTLS1_READ_BYTES:tlsv1 alert unknown ca]
#159 opened by GoogleCodeExporter - 6
No Ciphers Available when attempting WSS between SipML 1.3 / 1.5 and WebRTC 2.6.0
#157 opened by GoogleCodeExporter - 0
- 2
- 1
webrtc2sip crash
#154 opened by GoogleCodeExporter - 2
webrtc2sip configure failed
#155 opened by GoogleCodeExporter - 3
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 by GoogleCodeExporter - 0
- 0
Ability to configure the SIP Proxy (IP and port) from webrtc2sip server configuration file
#151 opened by GoogleCodeExporter - 1
random crashs
#150 opened by GoogleCodeExporter - 1
Disconnected/Unautorized running siplm5 live demo with sip2sip recommended
#148 opened by GoogleCodeExporter - 0
No Media (audio) When The Call Gets Connected to Voice Mail on a SIP Server
#149 opened by GoogleCodeExporter