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QOS Bearer/Signaling port

francisco_1
Level 7
Level 7

I am building a qos policy based on port ranges. I'm i correct in saying that RTP (range 16384 32768) is the voice bearer port and should be given dscp ef and RTCP,H225,H323,H245,SCCP are all singal protocols and should be give af31. If not what port i can give ef and af31 in my class maps?

6 Replies 6

andrew.prince
Level 10
Level 10

Franco,

The Cisco QoS SRND suggests:-

Voice - CoS 5 - EF

Call-Signalling - CoS 3 - AF31/CS3

HTH>

so are the ports i mentioned above correct for Voice and Call Singaling?

Voice is udp 16384 - 32767

H.323 uses tcp 1719 and 1720 for signalling, dynamic ports for media

SIP uses tcp 5060 for signalling, dynamic again for media

SCCP uses tcp 2000 - 2002 for signalling

For the rest you need to check the RFC's - as off the top of my head I am not sure.

HTH>

As Andrew says the QoS SRND suggests RTP use CoS 5 and DSCP EF (46), signalling should use CoS 3 and DSCP CS3 (24). Older Cisco equipment may set CoS 3/DSCP AF31 (26), however you can in theory use what you want.

What I think you are asking though is what layer-4 ports will be used so you can capture this with ACLs and then use these in your class-maps. Unfortunately this is protocol dependant.

With Cisco SCCP the signalling is TCP and the IP Phones use a destination port of 2000, the RTP Media uses dynamic source & destination port range between 16384 32767. With other protocols (H.323, SIP etc) the layer-4 port numbers are different. H.323 uses various ports for different elements of the signalling (H.225, H.245), SIP again uses a different port for signalling (UDP or TCP 5060).

With regards to layer-4 ports for RTP I have found that different vendors implement different ranges - for example Ericcson (now Aastra) IP Phones use a narrow UDP source range between 17000 & 17013.

HTH

Andy

thanks guys.

I have rated you as well.

Francisco.

Hello Francisco

The following document can help you, with all the port numbers between call managers, voice gateways, ip phones etc...

Hope this helps, if not too late :)

I used this to develop my qos design , for voice side..

Raj

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