[Asterisk-Users] Dealing with bandwidth limitations

[Bruce Komito]

We have a number of sip-connected customers whose broadband connections
have suddenly become, uh, less than reliable.  Actually, there is nothing
wrong with the broadband connection, but rather the network backbone in
the country they are connected through has become bogged down.  Although
latency between the sip clients and the * server is only 125ms (ping
times), it seems larger packets either take longer or get lost completely,
and the resulting latency as reported by * is 500-2000ms.  The result
is broken up sound at one end of the connection.  (The other end is
fine, but that's probably because the routing between the * system and
the sip clients is asymetrical, so the problem apparently exists in one
direction but not both.)  The sip clients all use G.729.

My question is this.  Are there any RTP settings that I could tinker with
that would improve the quality, perhaps at the expense of delay, by making
better use of the limited bandwidth available.  The problem is not so much
that the bandwidth is limited, but that it is intermittent and
inconsistent.

TIA

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815