There’s actually *no* voice traffic going across port 5060 at all. Not a schmick. Nada. Port 5060 is the SIP port, correct, but SIP means Session Initiation Protocol, which is where one end says “Hey, good lookin’, whatchya got cookin’?” and the other replies with “Some goose here wants to call some idiot on your end. Are you up to it and can you be arsed taking the call?” to which the other end replies “Sure, why not – I’ve just had a relaxing weekend off. Tell ‘em to use g.711 alaw and send their voice data through port 13294 and I’ll see what they want.” Or words to that effect.
Basically, SIP traffic is a simple handshaking that goes on before and during an RTP session. That’s all, so it uses what’s technically referred to as “bugger all” bandwidth.
Now, the RTP traffic, which contains the meat part of this sandwich, uses other ports, generally between 10,000 and some higher number. These port numbers are defined by the VoIP system you’re using. It is these ports that need to have QoS applied to them as it is these ports that contain the critical data such as “Darl, when you come home tonight, you need to bring some milk.” And “OK, love, anything else you need, like a big manly hug?” “No. That’s not going to work tonight either.”
So, if your router and network can apply QoS, it needs to be applied to the right ports/traffic, however if they are sharing this Internet connection with their data, don’t expect miracles (unless they are next to the telephone exchange). Also you need to remember that 99% of the data that is QoSed inside your network loses that QoS when it hits the ISP’s network, so you can really only QoS a) outbound traffic until it gets past the gateway and b) inbound traffic *after* you’ve already received it.
Stay tuned for more exciting episodes of VoIP and QoS when either a) I can be arsed writing some or b) you ask for more.
Regards,
The Outspoken Wookie
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