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What Protocols does Avaya VOIP use? (For traffic Shaping purposes)

Posted on 2009-04-22
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I have a setup running running an Avaya IP Office setup, and 4 remote VOIP [5602SW+ IP] telehones. These work fine, however when teh ADSL line that carries the traffic into the main site get busy we start to get drop out on the audio. I am therefore thinking of defining a traffic Shaping (Packet shaping) Policy for our Gnatbox 250 firewall, in order to ringfence 40-60kbs for VOIP. However beyond (possibly) SIP (UDP 5060/1) I am not sure what protocols this setup might utilise? Although the IP Office system can utilse H323 we dont have a gateway defined and therefore I dont think that we make use of it, I am not sure as to whether this setup uses IMS. I think that the system may also use RTP/RTCP when making calls too but I am not sure about this. The supplier of our telephone exchaneg doesnt seem to have any idea about this, and so far the internet has not been too helpful.

Any thoughts and suggestions would be very helpful.

Thanks
Paul

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Question by:bmtech
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denisdsr20 earned 200 total points
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Here are basic tips on VOIP :

SIP handles the session Submit calls, clear, redirect, allow phone to register. SIP default port is 5060 on UDP, is rather short message some 1000 of bytes par call, and you do not need to speed up SIP connections.

The media part (voice a/o video) is handled in RTP, it comes in UDP packet at rather high rate (about 1 packet each 20ms for voice, each packet is from 20 to 160 bytes depending on the codec used G771 Alaw, G729, ...). So any issue in voice quality comes from the RTP flow, this one should be handled as fast as possible. Take into account the hich rate and the fact it is short frames.

In order to adapt the flow between both side we often use RTCP, it sends reports about the RTP flow statistics in order to tune with best "effort" the RTP flow on each side. RTCP is some 100 bytes per second.

RTP flow uses 2 "channels" per call one to send audio the other to receive. It uses UDP in range from 20000 to 30000 but any range could be configured on sip proxy or on sip phone. And generally the RTCP is just one over the RTP. (if RTP uses port 30001, you may find RTCP at 30002).

Some tips about the bandwith required for audio only :
- in G711 - Alaw each channel is 64kb/s this means 2*64Kb/s for a call.
- in GSM each channel is 8kb/s (2*16kb/s for a call)

If you enable VAD (voice detection) silences (blank audio) may not be send so the requirement in network bandwith decrease.

Hope this helps

Denis DIDIER
SR20 service / France

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by:bmtech
ID: 31573271
Hello Denis,
This is excellant. I had figured out some of this yesterday. However the information on how RTP works has enabled me to complete the setup. I have examined the setup in the IP Office system now I know what to look for. It seems that by default the SIP Proxy/RTP range is from 49152 to 53246, this has enabled me to define System Objects in the GB250 to process that port range for the IP address of our PABX, and thereby force that into a packet shaping policy. I have started out with a band width of 80kbs, although we have 4 phones on this circuit they are not usuallly in use concurrently. Some information I found suggests 40kbs is sufficient per conversation, but I will see how it goes.

Best Regards, Paul
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