Kamailio SIP Server Wiki

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====== RTPProxy-NG ======

This page is intended to collate suggestions for improvements and new features for RTPProxy and the Kamailio RTPProxy module.

==== Architecture ====

=== JSON Interface ====
  * Will make the interface simpler and easier to extend.

=== Send/receive SDP to/from RTPProxy ===
  * Enables the RTPProxy to see and directly manipulate the SDP.

==== Kamailio RTPProxy Module ====
=== re-INVITE/UPDATE failure handling ===
  * When a re-INVITE/UPDATE fails the original stream should be used.
  * Currently, the module closes the old stream.
  * Also may cause problems with early media where early stream and active stream are connected to different end-points.

=== Hangs during startup when RTPProxy instance is unavailable ===
  * Each RTPProxy instance is contacted in-sequence.  If one or more RTPProxy instances is unavailable Kamailio hangs until the connect attempt times out.
  * RTPProxy connection at start-up should be asynchronous.

=== Enable/disable and add/remove RTPProxy instances ===
  * Would be good to load sets from a DB.
  * DB should contain indications whether RTPProxy is enabled/disabled at load.
  * Reload DB to change RTPProxy sets (and enable/disable proxies within sets) - probably needs hash-table data-structure changes.
  * Be able to mark RTPProxies enabled/disabled through MI command.  Existing calls (and re-INVITE/UPDATE) on disabled proxies should continue to work, but new calls should not go to a disabled proxy.

=== Receive RTP timeout notifications ===
  * rtpproxies should be able to notify, via the new rtpproxy protocol, media timeouts to the corresponding Kamailio instance (to the rtpproxy module).
  * The rtpproxy module should be able to trigger some event route in which the user could decide to send a BYE in both directions, stop the accounting or whatever. No more HTTP XMLRPC for notifying timeouts to a single Kamailio server (hack).
==== RTPProxy ====
=== ICE Support ===
  * TURN Server as well as RTPProxy
  * Handle ICE candidate lines in SDP
  * Breaking ICE (for CALEA and call recording)
  * Architecture, SDP sent to/from RTPProxy (from above) will be required

=== Bridging ===
  * Currently can only specify two networks
  * Would be good to be able to specify many networks and bridge between any combination

=== Failover ===
  * Deploy RTPProxies in pairs, replicate media sessions between them, fail-over/swap between instances.

=== RTP timeout notification ===
  * If a media sessions timeouts (so rtpproxy closes its sockets) rtpproxy should notify it to the proper Kamailio instance by providing data enough to Kamailio about the session (call-id, from, to...).

=== RTP/AVP to RTP/SAVPF conversion ===
  * For WebRTC
  * Need to be able to decrypt traffic from WebRTC and encrypt traffic to WebRTC
  * Need to be able to terminate and originate RTCP messages for the stream from/to WebRTC (if the other side does not support WebRTC)
  * ICE support (above) will be required

=== SRTP-RTP Gatewaying ===
  * Works only if RTPProxy sees the session key (e.g. SDES: RFC 4568)

=== Session Statistics ===
  * The current statistics give little information about the media session. The most obvious missing informations is the codec used during the session.