Open Voice Platform

Monday, April 21, 2008
10:26:00 PM EDT We have opened our Voice Platform to the world for the first time to enable any Open SIP Device or  third-party software client to integrate with our AIM CallOut Service .

Our Open SIP Gateway is compatible with a wide range of SIP compliant devices, including Mobile handsets, PC based softphones, Analog Telephone Adapters, IP Telephones, VoIP enabled Wi-Fi handsets and VoIP enabled open source and proprietary PBX'es.

Developers are encouraged to build or test any application they wish that is consistent with the Call Out Terms of Service.  Any SIP stack that strictly complies with the following standards should be capable of working with the AIM Call Out SIP Gateway.

 

• RFC 3261 – Session Initiation Protocol
• RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
• RFC 4028 - Session Timers in the Session Initiation Protocol (SIP)
• ITU-T E.164 Formatted Telephone Numbers
• RFC 2617 - Basic and Digest Access Authentication

Note: We currently do not support RFC 2069 - Digest Access Authentication since its not secure as RFC 2617.

For more details on on our Open Platform ,visit our Voice Portal at  http://voice.aol.com.

For more technical information , pls visit  http://dev.aol.com/aimcall



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