Peer-To-Peer Dialing - Matrix Telecom SARVAM UMG System Manual

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Peer-to-Peer Dialing

Making an IP call without the intervention of a proxy server is called Peer-to-Peer Calling. As Peer-to-Peer calling

does not require a proxy server, voice communication using this application can be done virtually free of cost. The

major cost savings offered by this application makes it a very attractive mode of inter-branch or intra-office voice

communication.

Let us understand how to use Peer-to-Peer Calling with the following illustration.

Two offices are connected to the IP network.

At Location A, a PBX (PBX A) and a Gateway (SETU VFXTH) is installed as shown above.

SARVAM UMG is installed at Location B.

Peer-to-Peer calls can be made between the two locations with suitable configuration of SARVAM UMG

and the Gateway (SETU VFXTH).

At Location A, you need to do the following configuration in SETU VFXTH:

Select a SIP Trunk to be used for this application and enable it. For example, SIP Trunk 1.

Set the SIP Trunk Mode of this trunk as Peer-to-Peer.

Keep the SIP ID of the SIP Trunk blank.

In the Router, you must configure the same SIP and RTP Ports as configured in the SETU VFXTH. In other

words, you must configure Port Forwarding for SIP and RTP on the Router.

By default, Allowed IP Address for Incoming SIP Message is set to As per Peer to Peer table. In

the Peer to Peer table at Location B, you must configure the IP Address of the Router at Location A.

Under Handling of Incoming Calls on the SIP Trunk, select the Incoming Call Routing option as Route

all incoming calls (with CLI) - to the Called Party Number.

Matrix SARVAM UMG System Manual

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