21 Advanced Media Termination Support
The Oracle® Enterprise Session Border Controller (ESBC) supports VoIP calls through the browser-based, real-time communication known as Advanced Media Termination. Using W3C and IETF standards, Advanced Media Termination supports cross-browser video calls and data transfers, such as browser-based VoIP telephony and video streaming. Advanced Media Termination allows users to make and receive calls from within a web browser, relieving the need to install a soft phone application. With Advanced Media Termination, the ESBC can enable users to communicate concurrently with one or more peers through various browsers and devices to stream voice and data communications in real-time through a variety of web applications. Advanced Media Termination also supports communications through end-user clients such as mobile phones and SIP User Agents.
- connected to networks with different throughput capabilities.
- on variable media quality networks (wireless).
- on fire-walled networks that don't allow UDP.
- on networks with NAT or IPv4 translation devices using any type of mapping and filtering behaviors (RFC 4787).
Supported Advanced Media Termination Services
- ICE-STUN (Lite mode) - Interactive Connectivity Establishment - Session Traversal Utility for NAT (ICE-STUN) enables an Advanced Media Termination client to perform connectivity checks. Use ICE to provide several STUN servers to the browser by way of the application. ICE processing chooses which candidate to address. Other benefits include support for IPv4, load balancing, and redundancy. ICE STUN support requires configuring an ICE Profile and specifying the profile in Realm Config. See "Configure ICE Profile" and "Configure Advanced Media Termination in Realm Config."
- RTP-RTCP multiplexing - Enables Real-Time Protocol (RTP) and Real-Time Control Protocol (RTCP) packets to use the same media port numbers. RTP is used for real-time multimedia applications, such as internet audio and video streaming, VoIP, and video conferencing. RTCP is used to monitor data transmission statistics and QoS, and helps to synchronize multiple streams. RTP-RTCP support requires enabling RTCP Mux in Realm Config. See "Configure Advanced Media Termination in Realm Config."
- SIP services including codec renegotiation, late media, early media, PACK interworking, attended and unattended call transfer, call forking, music on hold, transcoding, and High Availability.
Supported Protocols
- IPv4 for signaling and media
- UDP-RTP and UDP-RTCP on media
Supported Codecs
- Silk, OPUS, G.729, and G.711