Depends a lot on your setup.
If you are running e.g. an Asterisk server, you can
- prioritize all traffic to/from the Asterisk server IP number
or
- Asterisk (and most SIP clients) allows you to specify which UDP port
numbers to use for the RTP data. Proiritize traffic to/from this port range.
I know of some sites that run an Asterisk SIP proxy mainly/only to make
it easier to prioritize the VOIP traffic.
or
If you are using hardware VOIP phones, put them in a specific IP range
and prioritize this range.
or
Many hardware phones and some software VOIP clients support setting QoS
flags in the data packets which both switches and routers can use to
prioritize the traffic. This can be at layer 2 (e.g. 802.1Q / 802.1p) or
layer 3 (DiffServ, IP ToS)
As mentioned before, SIP is easy (almost always on port 5060), it is the
RTP data stream that can be tricky.
My experience: if you control the infrastructure, the easiest and
cheapest way to ensure good VOIP quality is to often to make sure there
is _plenty_ of bandwidth. This is seldom a problem on the LAN, but may
be a problem on your internet connection if you do not own the
infrastructure.
**
sincerely
Nicolas Padfield
Beat Meier wrote:
Hello
How can I filter (i.e. priorize) RTP protocol and SIP?
Has anybody wrote a filter for that in the meantime
(In 2006 there was none answer from the list ...)
Thanks
Beat
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