[asterisk-dev] limited packetization period for gsm codec?
wojtek at puchar.net
Wed Jul 29 04:12:23 CDT 2020
i have asterisk 1.8 on one side and older 1.3 on other side.
the other side is MIPS based router with OS replaced to FreeBSD 10 - hard
to upgrade, as it was quite difficult already to make asterisk work on
But works fine. The problem is that i cannot have more than 100ms
when i set in sip.conf on both sides
it works fine. tcpdump shows 10 packets sent from each side. sounds
when i try gsm:200 i'm getting chopped sound and tcpdump shows definitely
something wrong is going on. lots of short packets but no errors reported
on either side.
Documentation says 300ms is allowed. I want long packetization period to
reduce overhead of IP/UDP/RTP as well as openvpn.
I want to limit amount of bytes transmitted as much as possible. 1/5s
delay is completely acceptable as long as sound quality is OK.
my sip.conf lines are:
other side is the same just IP address is different
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