Gateway SRTP

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Aug 23rd, 2007
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After enabling SRTP on our H.323 gateways, we can no longer dial a DID number for a specific user and get their voice mail. When using Call Viewer on the Unity server, you can see the call come into Unity just fine. All you hear is a busy signal.

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htarra Wed, 08/29/2007 - 14:14
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Try enable DID on incoming dial peer.

Media encryption using Secure Real-Time Transport Protocol (SRTP) delivers protection by encrypting the voice conversation, rendering it unintelligible to internal or external eavesdroppers who have gained access to the voice domain. Designed for voice packets, SRTP supports the Advanced Encryption Standard (AES) encryption algorithm and is an IETF RFC 3711 standard.

As long as any third party endpoint supports RFC 3711, things should work fine.


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