[LARTC] Filtering RTP/SIP protocol (Voip)?

[Nicolas Padfield]

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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