Hello.

I have a customer wishing prioritization of VoIP packets. They are using SIP protocol. They tell me it works like this:

UDP 5060-5063 used for call setup
UDP 16384-16482 used for call voice packets

I have set packet mangle to mark all UDP packets in 5060-5063 and 16384-16482 with flow mark "VoIP".

I have added queue trees on global-in and global out to set VoIP market packets to priority of 1.

I see traffic in 5060-5063, but no traffic in 16384-16482 at all. When I set packet mangle to mark all traffic (not just UDP) in 16384-16482, I see traffic.

I think this is because the packets are really RTP, which uses UDP but might have a different packet identification.

Can someone comment on this?

Is it possible to mark RTP packets in Mikrotik?

Is anyone else doing QoS for VoIP?

Hi.
Would you send mangle teble.
I have a quesione about what type of queue you are using for voice? I have a problem with interuptions during conversation.
Kamen

UDP 5060-5063 used for call setup

SIP uses 5060 but not necessarily

UDP 16384-16482 used for call voice packets
I have set packet mangle to mark all UDP packets in 5060-5063 and 16384-16482 with flow mark "VoIP".
I have added queue trees on global-in and global out to set VoIP market packets to priority of 1.
I see traffic in 5060-5063, but no traffic in 16384-16482 at all. When I set packet mangle to mark all traffic (not just UDP) in 16384-16482, I see traffic.
I think this is because the packets are really RTP, which uses UDP but might have a different packet identification.

yes rtp uses udp but you can't tell it's rtp (without some heuristic analysis like ethereal do) just simply looking at the packet -- router (at least MikroTik) will see it as udp

Is it possible to mark RTP packets in Mikrotik?
Is anyone else doing QoS for VoIP?

applications which use rtp for payload transmission agree what ports will they use during session initiation (via sip/sdp headers), rtp can use any port from 5000 and higher
usually but, again, not necessarily rtp uses ports somewhere in 15000-35000, most applications however allow setting rtp listening port
for example Xlite (sip application from xten) uses rtp port 8000 by default
so! you have to know which ports your customer uses for sure and mark them
you also may want to set new TOS (actually it is Differentiated Services Field) for VoIP traffic, say 48 (priority, low delay), when marking packets (set-tos=48)

Hi Roman.
Do you have proposals about my problem with queuing ?
Kamen