diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 62af0e21380d5703a199af1ad23165f1508ee37e..287e3c52be76da3e8d0467d35215847e609346e4 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -255,6 +255,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Message-Account in the MWI notify message ; defaults to "asterisk" +; Codec negotiation +; +; When Asterisk is receiving a call, the codec will initially be set to the +; first codec in the allowed codecs defined for the user receiving the call +; that the caller also indicates that it supports. But, after the caller +; starts sending RTP, Asterisk will switch to using whatever codec the caller +; is sending. +; +; When Asterisk is placing a call, the codec used will be the first codec in +; the allowed codecs that the callee indicates that it supports. Asterisk will +; *not* switch to whatever codec the callee is sending. +; ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer.