Configuring SIP on WebRTC (WSS)

On a default FreeSWITCH installation you only need to edit the "internal" SIP profile, in /usr/local/freeswitch/conf/sip_profiles/internal.xml. You want to check "wss-binding" for the TCP port where SIP (mod_sofia) will listen for signaling on WSS transport.

  <param name="wss-binding" value=":7443"/> 
  <param name="ext-rtp-ip" value="93.58.44.181"/> 
  <param name="ext-sip-ip" value="93.58.44.181"/> 

The values of ext-rtp-ip and ext-sip-ip sets the IP address SIP will tell the WebRTC clients they must connect to in order to exchange signaling and media streams. It must be set to FreeSWITCH IP address as seen from the WebRTC clients. So for clients coming from the Internet, ext-rtp-ip must be set to the external side of the NAT, eg to the routable IP address (often same as the webserver IP address).

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