Voice & Video Services
Explore the technical features of our voice & video services. Click on each feature to read more.
- The service hosts registrations of devices and the internal system is connected via a SIP trunk
- Proxy integration where the service relay requests originating from clients on the internet to the internal system
Since media is supposed to flow end-to-end between SIP devices, the SIP Server will not be part of the media flow at all in some cases. When media anchoring is required (e.g. to apply Hosted NAT traversal) the media streams are relayed without any intervention and media remains encrypted end-to-end.
SIP devices must support SRTP (RFC 3711) together with an appropriate key negotiation method to support end-to-end RTP encryption (SDES RFC 4568 or DTLS-SRTP RFC 5764).
Talk to us today or visit our Developer Area to learn more about our Voice & Video APIs.
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