[asterisk-users] "internal_timing" not working (re: SIP silence suppression)

[Benny Amorsen]

David Backeberg <dbackeberg at gmail.com> writes:

> As I understand it, you have enabled silence suppression, the silence
> is getting suppressed, and that is worse than when you were not using
> silence suppression. So how about not using silence suppression?

Silence suppression isn't just a "break your telephony" option, it
actually has a purpose. It can often save half your RTP bandwidth at
close to zero cost in quality.

It would be nice if it worked well in Asterisk. (Not that I can say
personally whether it works; it's been 3 years since I tried it).


/Benny