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Solutions
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VoIP Freedom for $37 per port What do you do when an important customer has a booming global VoIP network, but no price effective way to extend the multi-lateral VoIP peering of its wholesale network to the vast market of enterprise customers? This issue is exactly the challenge TransNexus faced when one of our carrier customers asked us to identify a cost effective solution for providing enterprise VoIP customers with secure access to the carrier’s managed wholesale VoIP network. The Challenge Our customer’s global wholesale VoIP network is growing rapidly and they are eager to add high margin VoIP traffic from enterprise customers directly to their wholesale network. The problem, identify a VoIP platform for small business customers which meets the following requirements. · Can be a VoIP gateway for customers who already have a traditional PBX. · Can be an IP PBX. · Scales from four analog lines to four E1/T1 connections. · Supports the OSP peering protocol for secure access control and call detail record collection with the carrier’s global wholesale network. · Offers total cost of ownership that is affordable for a small enterprise. Finding a VoIP platform which satisfied all requirements was a challenge. There are VoIP platforms which satisfy the technical requirements, but they cost $200 or more per voice channel. The Solution The search for a VoIP platform which makes economic sense for small enterprise customers led to the growing field of open source VoIP solutions. After several months of research and testing we identified the AsteriskTM Open Source PBX (www.asterisk.org) as an excellent solution for our customer. We selected Asterisk for the following reasons:
How Multi-lateral VoIP Peering Works Multi-lateral VoIP peering is managed by a trusted third party, or carrier, who is the central routing and access control authority for a large number of anonymous peer VoIP networks. Each VoIP network has a bilateral interconnect agreement with the central trusted third party, or carrier. As the central certificate authority, the carrier is able to provide route discovery and secure inter-domain access control services for the anonymous peer networks that do not have interconnect agreements. Setup. Participation in a multi-lateral peering network requires that each Asterisk platform exchange public keys with the certificate authority (CA) or OSP server. In addition, each Asterisk device must generate its own certificate and have it signed by the CA or OSP server. The OSP client support in Asterisk provides these features. Call Scenario. The diagrams below show two Asterisk networks and describe the simple multi-lateral peering call flow. The phones managed by Asterisk may be SIP phones on the IP network, local analog phones connected via an FXS interface or phones connected through the PSTN. 1. A calling party in the green network originates a phone call. If the called party is part of the green network, Asterisk will complete the call directly. However, if the called party is not part of the green network, Asterisk can send a Peering Request to its multi-lateral peering service provider. The OSP peering server authenticates the Asterisk platform and performs a route and access permission lookup based on the called party’s e.164 telephone number, tel uri or sip uri. 2. If the calling VoIP peer network is authorized to interconnect with the destination VoIP peer network, the OSP peering server returns an Peering Response with the IP addresses of destination peer networks which can complete the call. Included in the response is a digitally signed token authorizing the call session.
3. The source Asterisk proceeds to send a SIP INVITE message directly to the destination Asterisk in the red peer network. 4. The destination Asterisk in the red network receives the SIP invite from the unknown peer. Before rejecting the INVITE, the destination Asterisk validates the digital signature of the authorization token using the public key of the multi-lateral peering service provider. If the token is valid, access is granted and the call is routed to the called party.
5. When the call is completed, both the source and destination peers send a call detail record, based on the OSP standard format, to the OSP peering server. For more information on Asterisk see www.asterisk.org. Asterisk and Digium and trademarks of Digium, Inc. |
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