Hello !
Are there anybody around with experience with QoS for SIP on RouterOS ?
How do you mangle the datastream (udp on a wide range of ports)
Thanks in advance
KimC
I am currently trying the same thing! I have already prioritzed the ports that are documented by connection to the SIP providers server. But I think SIP uses many more ports from what I have looked at so far in the open connections when the call is made.
Do you have any knowledge of what ports SIP uses overall?
SIP normally uses 5060, but doesn’t have to and in fact can use anything it really wants. RTP (the protocol carrying the audio) normally uses 10000-20000 but, as with SIP, can use anything it wants.
The trick is to stop trying to use ports to identify the VoIP traffic. We use TOS bits and server addresses very successfully.
The master proxy port and the SIP proxy port can probably be changed in the phone, but the RTP port may be hardcoded. I have a wireless Zyxel SIP phone and the RTP ports used by the phone is port 2070 - 2073. There is no way to change it as far I know, and since I know the port numbers, I can easiley set up rules for QoS. I have not set up QoS for it in a Mikrotik router because I still use another solution right now built on a linux server, so I cannot give you a configuration example I for sure know that will work out for you.
Yeah I guess your right, but it does mean tying up with certain providers or getting info from end users about what voip services they are using. It would be nice to have a sort of stateful tracking of voip connections like with the p2p traffic module!
True, but we only QOS to our VoIP servers. Traffic to any other providers VoIP servers is just normal stuff to us and we don’t care. If a customer paid more to have QOS on VoIP to an outside provider, then we would be happy to provide that, but not at the normal rates.