Enabling direct RTP streams between SIP phones in Asterisk

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

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