System information

 RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging
 RFC 1889 RTP: A Transport for Real-Time Applications
 RFC 2190 RTP Payload Format for H.263 Video Streams -only routing
 RFC 2327 SDP: Session Description Protocol
 RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
 RFC 3264 An Offer/Answer Model with Session Description Protocol
 RFC 3550 RTP: A Transport Protocol for Real-Time Applications -replaces RFC 1889
 RFC 3555 MIME Type Registration of RTP Payload Formats
 RFC 3911 The SIP "Join" Header
 RFC 3324 Network Asserted Identity
 RFC 3326 The Reason Header Field
 RFC 3581 Symmetric Response Routing
 draft-ietf-mmusic-ice-02 A Methodology for NAT Traversal for Multimedia Session Establishment Protocols
 draft-ietf-avt-rtp-ilbc-04
 draft-ietf-sipping-cc-transfer Call Control - Transfer
 draft-ietf-sip-referredby-05
 Custom protocol extensions are possible
1.2.3. Codecs
 G.723.1
 G.729
 G.711 A-law
 G.711 u-law
 GSM 06.10
 GSM
 Speex 2,3,4,5,6 (narrowband, wideband and ultra-wideband)
 G.726 (16,24,32,40 KHz)
 G.722
 T.38
 DTMF
 Custom 1 kbits codec
 All other codec’s for pass-trough
 Voice:
• Adaptive de-jitter buffer
• Voice Activity Detection/Silence Suppression
• Recording conversations (In Stereo caller/callee left/right)
• QoS