[asterisk-dev] RTP trunking - 58% savings on media bandwidth?

[Maxim Sobolev]

I looked into this technology a while ago, while working on 
re-packetization feature for the RTPproxy. The concept is certainly 
interesting one, however the big problem is that both ends of a "trunk" 
should support the same protocol and I believe there is no standard way 
to negotiate it in the SDP. In many cases you can get sufficient saving 
by just increasing RTP payload size. By going from 10ms to 90ms you get 
about 70% saving on G.729 and it works with any standard UA, no 
additional support is required. Subjectively, sound quality at 90ms is 
acceptable (at least if your packet loss rate is not very bad).

Regards,
-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
T/F: +1-646-651-1110
Web: http://www.sippysoft.com
MSN: sales at sippysoft.com
Skype: SippySoft