Site without gateway: WAN bandwidth during transfer

[rime]

When I execute a transfer on an IP phone at a very small site without Media Gateway, is there just one incoming RTP flow in progress during the whole transaction or is there a point when 2 incoming RTP flows coexist even if only one is heard at the IP phone?
Example:
- I have a call in progress, one incoming RTP flow is active through the WAN interface for my call. There is an outgoing RTP flow too, of course.
- I press the transfer button so I now hear dial tone coming from a Media Gateway resource at another site. The other party hears Music-on-Hold, also coming from some Media Gateway resource at another site. But what happens to the RTP flow from the other party? Is it simply not existing anymore at the WAN interface of my site without Gateway or do I still have to account for it in bandwidth calculations?
Now the other way around: what happens to the RTP flow that my IP phone was sending across the WAN interface before I hit "transfer": is it no longer existing or is it still sent, possibly to the Media Gateway which generates the dial tone? I am still talking about the moment after I hit "Transfer", while hearing dial tone, before dialing the transfer destination.

I would assume that the RTP flow which is not supposed to be heard ceases to exist, but I have no certainty. The issue is that it does have an impact on how much real-time voice bandwidth I must reserve at very small sites without Media Gateway. For instance, I would book enough bandwidth for 5 parallel calls (and set CAC=5) at a site with 4 users to cope with this possibility.

The same discussion applies for initiating conferences or swapping between 2 or more lines with active calls. Do I always have just one incoming RTP flow or can as many flows exist as lines waiting + the active one?

Alternatively, is there a set of recommendations about how to cope with very small sites without gateway in terms of real time WAN bandwidth? Thanks in advance!

[aa1]

direct vs. shuffled calls

rime, this is a very interesting topic and one that is quite complex as well.

The discussion is very much dependent on the call flow, the nodes involved in the call. the number of parties involved and the administration. Let's look at two scenarios:

- direct IP
- in-direct IP

In the case of direct IP, then two voip resources (let's say two IP phones) will transmit RTP directly to each other. bypassing the VoIP resources in the system (MedPro, mediageway...). in the case of in-direct IP, the RTP stream goes through some sort of VoIP engine.

In the case of a remote site with an IP phone (H.323), the phone would either talk to a VoIP resource or to another IP phone. (depending on the administration). The RTP stream would be between the two nodes.

The phone is assigned a network region. This may be done via the gatekeeper, or ip-network-map form. Nevertheless, the phone has a network region. When you pick up the handset, the phone receives dialtone from a VoIP resource in the same network region as the phone. that stream comes from that VoIP engine. Then when you phone initiates a call, CM may shuttle the VoIP resources (i.e., ask the phone to talk to another voip engine or directly to a phone), so the stream moves from one resource to another. Of course, with the help of administration., you may force a phone to always talk to set of media gateways regardless of the call flow.

The administration on the CM is quite important to modify the way you would like the call to work and resources to be utilized.

[aa1]

Avaya Communication Manager Network Region Configuration Guide

A good source of information:
Avaya Communication Manager Network Region Configuration Guide

https://downloads.avaya.com/css/P8/documents/100011997

[aa1]

still have to account for it in bandwidth calculations

Question you asked:
************************
- I press the transfer button so I now hear dial tone coming from a Media Gateway resource at another site. The other party hears Music-on-Hold, also coming from some Media Gateway resource at another site. But what happens to the RTP flow from the other party? Is it simply not existing anymore at the WAN interface of my site without Gateway or do I still have to account for it in bandwidth calculations?


Communication manager may shuffle the RTP stream during the call. (this does not mean that CM continuously shuffles the stream when phone A and phone B are talking to each other). It means it may shuffle the call (RTP stream) when there is the need. So in your example, when the transfer button is pressed by phone B, the call is put on hold. phone B's first call appearance is put on hold and second call appearance is lit and phone B hears dial tone. Now, the dial tone comes from a tone generator and it is provided to the phone B through a voip resource over RTP stream. The phone B may have been receiving the RTP stream from another source, but now, it receives it from either the same source or another source. The codec with which the dial tone is sent to the phone (in the RTP) comes from the network region to which the phone belongs to. (this is important for your bandwidth calculation). However, the phone B might have been talking to phone A using another codec set (hence a different bandwidth).
The phone A now hears the MOH from a VoIP resource via RTP stream. so the RTP stream between phone A and phone B (assuming the phone A and phone B were talking directly to each other) will stop and a new RTP stream will start from the MG to the phone B to deliver the dial tone and from MOH source to the phone A to deliver the MOH.
You still do have to consider and calculate the bandwidth

[aa1]

phone to talk to MG all the time

One of the methods would be to configure the remote site phones to always talk to a MG in a near site. This way the phone will always communicate with the VoIP engine in that MG. you may then assign a codec set of your desire. you may then use that codec set in your bandwidth calculation to estimate the amount of bandwidth needed per call and then multiple that by the number of calls you wish to have simultaneously.

[aa1]

Wireshark

You may also capture a network trace using wireshark from the phone side and observe the RTP communication between the phone and the VoIP resources. Naturally the outcome depends on the type of the administration on CM. this is naturally for learning purpose...

Hope these info will help and answer your questions/concerns.
Arbi