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
SIP is pretty easy. Normally it uses 5060 port. But prioritizing this port traffic won''t enhance the overall voice quality. RTP ports are decided dynamically during SIP handshake. To filter RTP protocol from packet pattern will delay the traffic. So using port number is easier way, BUT, you need to find out these ports from the SIP handshake messages. How to prioritize depends on your setup''s capabilities. -----Original Message----- From: lartc-bounces@mailman.ds9a.nl [mailto:lartc-bounces@mailman.ds9a.nl] On Behalf Of Beat Meier Sent: Friday, October 12, 2007 7:48 AM To: lartc@mailman.ds9a.nl Subject: [LARTC] Filtering RTP/SIP protocol (Voip)? 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 _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc
Salim S I wrote:> SIP is pretty easy. Normally it uses 5060 port. But prioritizing this > port traffic won''t enhance the overall voice quality. > RTP ports are decided dynamically during SIP handshake. To filter RTP > protocol from packet pattern will delay the traffic. So using port > number is easier way, BUT, you need to find out these ports from the SIP > handshake messages. > How to prioritize depends on your setup''s capabilities. >Correct. However, some conntrack modules are available for such dual port (negotiated data port) protocols like ftp. Need to see if one exists for SIP. If so, iptables can be used to mark SIP data connections and the mark can be used for traffic classification. Search netfilter.org mailing lists please. Mohan
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 > _______________________________________________ > LARTC mailing list > LARTC@mailman.ds9a.nl > http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc