p2p communication bet 2 sip client from a cyber cafe, HOW ?

[nisha.shukla]

How can p2p ( peer to peer) communication bet 2 sip client is possible , if both SIP client knows each other IP address.

Do we need to use the service( take help of) of SIP proxy server ? Can someone explain me the call flow or architecture of this , taking an example that I am using SIP client form a Cyber-cafe ? And also in what way MG( Media Gateway) will come into picture ?

[wmseto]

This is interesting concept. Let's discuss this...

Are you talking an application/client that will be created by you or someone? If so, I think this is possible. I imagine the SIP INVITE will be send from one client directly to another client, and the SIP ACCEPT is reponsed directly also. Therefore yes, you can skip the proxy. As far as the media streams (audio) they can be p2p also, usually these will be using RTP, I suppose.

If you are talking about the VoIP client that is shipped with S60 phones (i.e. N80, E60, E61 etc). I don't think it is possible to skip the SIP proxy server. It is not the protocal issue; it is just that the Nokia implementation always take into the mobility idea in mind, where IP address of a client cannot be fixed. Now, you can use some free servcies, such as Gizmo and Truphone. These servcies utilities S60 VoIP client and provide free SIP proxy services.

I cannot common on the Media Gateway too much. I have read though, Media Gateway can be used for rich call experience. Say sending ringtone to the peer for the call, or shareing other media with the call. That is the level I have understand it.

[amardeep123]

Talking on the same line, if with S60 phones (i.e. N80, E60, E61 etc). we cannot ommit SIP proxy, its means if a mobile changes its domain/networking while it is in a session/conversation, then this SIP Proxy( or some similar network element element ) will allocate new IP address ( or might do some mapping with old IP address to new IP address ).., then could be mid-call signaling is possible ?

And I guess if two SIP clients in a session, talking p2p, could their session be via some non-ip network, where in MGC and Trunking Gateway(MG) might acts as a bridge , as below:

A SIP client --> Router 1 ---------> Router 2 --------------->MGC ----> Router 3 -------------> B SIP client
( Network 1 IP network) ( Network 2 Non-IP net) | ( network 3 - IP network............)
|
v
MG

Can this be possible ?
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[wmseto]

I think it is up to the application level how to handle network changes. If the user roams to another network, the connection is likely dropped or dis-connected. It is up to the application to re-register and reconnect. On the voice level, if the re-connect can happen quick enough, I suppose the call interruptions can bring to minimial.

SIP session, in theory, does not limit the media streams to be on IP only network. However, you have also mentioned, signaling will need to be on IP. SIP is itself has been defined to be on top of IP, and I don't know otherwise.

Besides, if p2p talk in not over IP... then we are out of the VoIP scope, right?