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Question-answer

RTP packets are not sent every 20ms exactly with RTMP SIP Gateway. Can we reduce the jitter?

As you know Flash sends the audio data via TCP stream.

Can we use flashphoner client with Android smartphone?

Android Chrome and FF browsers support of WebRTC, it means the same browser client can be used.

What is Opus codec bitrate for WebRTC call?

For one WebRTC call we have 4 opus streams:

Does WCS support ptime=10 mode? Our gateway use the packetization mode. How can we setup this?

The ptime 10 ms is not supported.

Is inband DTMF supported? How can i setup DTMF tone signals?

There are 3 types of DTMF:


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