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var connectionOptions = { urlServer: url, keepAlive: true }; if (authToken) { connectionOptions.authToken = authToken; } else { connectionOptions.sipOptions = sipOptions; } Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session, connection){ ... }); |
re-INVITE support
Since build 5.2.1943 re-INVITE for outgoing SIP calls is supported. If a SIP PBX sends SDP in 200 OK response to SIP INVITE from WCS, the SDP will be applied to media codecs negotiation to pass media traffic between WCS and SIP PBX. No changes in WCS settings required.
Known issues
1. It's impossible to make a SIP call if 'SIP Login' and 'SIP Authentification name' fields are incorrect
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generate_av_for_ua=all |
6. IVR greeting plays not from beginning if caller makes audio+video call
Symptoms: there is a gap before IVR greeting starts playing if caller makes an audio+video call (for instance, using Phone Video example)
Solution: update WCS to build 5.2.1755 and reduce video frames generator start timeout
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generate_av_for_ua=all
rtp_generator_start_timeout=100 |
7. An excessive video transcoding to VP8 starts when making a call between browsers
Symptoms: a video traffic is receiving from SIP server in H264 codec, but video is playing as VP8 in browser
Solution:
a) add the following parameter to WCS settings
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profiles=42e01f,640028 |
b) add the following parameter if above does not help
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proxy_use_h264_packetization_mode_1_only=false |