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``` Webrtc2sip server crashes on unhold the call . Here I give the messages we are getting on console ... *INFO: State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE *INFO: State machine: x0000_Connected_2_Connected_X_iINVITE *INFO: tnet_ice_ctx_set_remote_candidates...

Configure --with-ffmpeg fails

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``` What steps will reproduce the problem? 1. Configure & compile ffmpeg cd /usr/local/src git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg cd ffmpeg git checkout n1.2 ./configure --prefix=/usr --extra-cflags="-fPIC" --extra-ldflags="-lpthread" --enable-pic --enable-memalign-hack --enable-pthreads...

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

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``` What steps will reproduce the problem? Calling to/from Chrome for Android. What is the expected output? What do you see instead? Error connecting audio stream. Connection working fine for...

``` What steps will reproduce the problem? 1. Architecture: Chrome/sipml(Client side) Asterisk GenesysSIP webrtc2sip Chrome/sipml (Agent side) 2. When agents log with webbreaker=false, everythink is fine, but we need also...

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

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``` What steps will reproduce the problem? 1. Attempt to register a sipML client to webrtc2sip 2.6.0 with DTLS/SRTP enabled. What is the expected output? What do you see instead?...

``` What steps will reproduce the problem? 1. make call to extension, where PBX refuses the outbound call What is the expected output? What do you see instead? call should...

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

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``` What steps will reproduce the problem? 1. set-up - 2 sipml clients webrtc2sip sip server (freeswitch) 2. send an IM from 1002 to 1003 after registration What is the...

``` What steps will reproduce the problem? 1.Not applicable as this is a minor feature request 2. 3. What is the expected output? What do you see instead? Currently, this...

``` What steps will reproduce the problem? 1.Call a SIP user who is not registered from a registered SIP user on sipml5 using webrtc2sip gateway on a SIP server 2.Call...

Disconnected/Unautorized running siplm5 live demo with sip2sip recommended

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``` Hello Sir, I am new to webrtc , I have build webtrc2sip on my ubuntu 12.04 LTS. Now I am trying to run a live demo with account created...