Web Call Server 5.2 - EN
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Overview

Since build 5.2.660, WCS supports sending and receiving audio and video tracks via the same connection while publishing or playing WebRTC stream. This feature works also for streams inside CDN. This allows to reduce media ports usage and to decrease server load. The feature is supported for both UDP and TCP transport.

RTP bundle is enabled by default, and will be used if client supports it. If some problem occurs while connection establishing, RTP bundle can be disabled with the following parameter in flashphoner.properties file

rtp_bundle=false

Known issues

1. When RTP bundle is enabled, the statistics still shows 2 busy ports per connection because media ports are reserved before SDP exchange is completed, and server detects how much ports client will use.

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