Information for VoIP service providers
If you are a VoIP/SIP service provider, you can join our service and start offering VoIP termination to Google Talk, MSN/Live Messenger and Yahoo Messenger users. You may also use our GTalk2VoIP gateway to provide (sell) phone numbers to our users.

Terminating VoIP traffic from IM users to your equipment:

  • You must be a voip service provider with the ability to accept incoming SIP or H.323 calls to your equipment and/or soft-switches. You must be able to reconfigure your equipment according to the needs of accepting calls.
  • Your equipment must accept calling and caller ids in E.164 format. It's possible to send a tech-prefix from GTalk2VoIP gateway.
  • Your equipment must support one of the following codecs: G.711 (preferrable), iLBC, G.729 or GSM.
  • Register your company as a VoIP service provider on our site.
  • Define one or more SIP/H.323 gateways (gateway's IP address, tech-prefix and supported codec).
  • Define E.164 destinations and rates (cost of calls routed to your gateways).
  • Test your gateways by sending calls from GTalk2VoIP using a special feature. While testing, you must be able to accept a call from GTalk2VoIP gateway and dial DTMF tones.
  • Testing is required to ensure your gateway is configured properly and accepts calls from GTalk2VoIP users.
  • Your rates must be defined in US dollars in per-minute basis.
  • Accounting must be made on per-second basis.
  • To all your rates we will add 50% of your initial call cost for each minute (GTalk2VoIP standard comission). Final rates displayed and charged to users include our comission.
  • Please, do not forget to contacts us by e-mail:

   


Offering VoIP service and DIDs to IM users:

  • Configure your equipment to route SIP calls for IM users to our gateway. Use the following mapping plan:
    • SIP calls to Google Talk user should be mapped to
      username%
    • SIP calls to MSN/Live Messenger user should be mapped to
      username%
    • SIP calls to Yahoo! Messenger user should be mapped to
      username%
    • SIP calls to AIM user screenname should be mapped to
      screenname%
    • SIP calls to ICQ user screenname should be mapped to
      screenname%
  • Configure your equipment to support _sip._udp SRV records when routing calls to GTalk2VoIP.
  • We do not obligate you to obtain our permission to use the gateway for offering paid or free services, though it would be good if you contact us before you start sending SIP calls. So, we can plan the load and resolve technical problems/issues.

  • Once you register, you can start using API for our partners to automate user subscribtion process, etc.
Calling software

We recommend to use Talkonaut for making calls:

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