By default, asterisk installations will instruct SIP phones to pass their media streams (RTP streams) through the asterisk server itself. That is to say, the RTP stream would look something like this: (phone 1) <-----> (asterisk) <------> (phone 2) However, for performance reasons, especially in non-NAT environments, it is preferable to have the RTP streams … Continue reading Enabling direct RTP streams between SIP phones in Asterisk
