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-rw-r--r--configs/sip.conf.sample12
1 files changed, 12 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 62af0e213..287e3c52b 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.