Brian Cuthie wrote:
Let's say that I have a call coming in to Asterisk through a TDM400P and
going out through SIP to someone on the Internet. Is there any
configuration option that would allow me to do silence suppression on
the RTP stream generated by Asterisk on behalf of the TDM400P connected
user? SIP phones allow me to do this easily, but I'd like to be able to
conserve upstream bandwidth on calls that don't emanate from a SIP phone
here at my location.
Asterisk SIP does not support silence suppression. In fact, using Silence
suppression on an inbound RTP stream will lead to problems, since Asterisk
takes timing from inbound RTP streams.
/Olle
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