[asterisk-dev] limited packetization period for gsm codec?

[Wojciech Puchar]

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 
this.

But works fine. The problem is that i cannot have more than 100ms 
packetization period.

when i set in sip.conf on both sides
disallow=all
allow=gsm:100

it works fine. tcpdump shows 10 packets sent from each side. sounds 
properly


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:
[wojteks]
type=friend
host=10.1.3.1
call-limit=1
qualifyfreq=180
disallow=all
allow=gsm:100

other side is the same just IP address is different