- RFC 2833 – “RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals” is supported.
- RFC 2976 – “The SIP INFO Method” is partially supported: the PortaSIP cluster is able to either resend INFO requests to a remote UA or extract DTMF information if a call scenario requires it.
- RFC 3261 – “SIP: Session Initiation Protocol” is supported with the following limitations:
- The SIP URL domain is ignored in incoming requests.
- For non-clustered solutions, in the case of a direct incoming connection from a remote SIP UA to a B2BUA (where the SIP proxy is not engaged in the SIP message exchange), only UDP transport protocol can be used. For the PortaSIP cluster, TCP and TLS transport protocols are also supported.
- Dialog forking is not supported when PortaSIP is a User Agent Client.
- RFC 3262 – “Reliability of Provisional Responses in the Session Initiation Protocol (SIP)” is fully supported.
- RFC 3263 – “Session Initiation Protocol (SIP): Locating SIP Servers” is partially supported with the limitation that NAPTR records are not supported.
- RFC 3264 – “An Offer/Answer Model with the Session Description Protocol (SDP)” is partially supported for a late offer/answer model.
- RFC 3265 – “Session Initiation Protocol (SIP)-Specific Event Notification” is supported in the PortaSIP cluster.
- RFC 3311 – “The Session Initiation Protocol (SIP) UPDATE Method.”
- RFC 3323 – “A Privacy Mechanism for the Session Initiation Protocol (SIP)” is partially supported.
- RFC 3324 – “Short Term Requirements for Network Asserted Identity and 3325 – Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks” are partially supported.
- RFC 3326 – “The Reason Header Field for the Session Initiation Protocol (SIP)” is partially supported.
- PortaSIP sends the CANCEL request with the Reason header when a call is accepted elsewhere:
- Reason: SIP; cause=200; text=”Call completed elsewhere”
- RFC 3327 – “Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts” is supported.
- RFC 3428 – “Session Initiation Protocol (SIP) Extension for Instant Messaging” is supported.
- RFC 3489 – “STUN – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)” is supported.
- RFC 3515 – “The Session Initiation Protocol (SIP) REFER Method” is partially supported with the limitation that for attended transfer PortaSIP does not update the original CLI when sending a request to the transfer target.
- RFC 3550, RFC 1889 – “RTP: A Transport Protocol for Real-Time Applications” are partially supported with the limitation that if the RTP proxy generates the media stream (the actual voice traffic), it does not relay the RTCP packets.
- RFC 3551 – “RTP Profile for Audio and Video Conferences with Minimal Control” is supported, with the following limitation:
- Not all encodings are supported.
- RFC 3581 – “An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing” is supported.
- RFC 3611 – “RTP Control Protocol Extended Reports (RTCP XR)” is supported.
- RFC 3711 – “The Secure Real-time Transport Protocol (SRTP)” is supported (PortaSIP passes encrypted packets between the phones and does not perform any encryption).
- RFC 3856 – “A Presence Event Package for the Session Initiation Protocol (SIP)” is supported.
- RFC 3891 – “The Session Initiation Protocol (SIP) ‘Replaces’ Header” is supported.
- RFC 3903 – “SIP Extension for Event State Publication” is supported.
- RFC 3951, 3952 – “Internet Low Bit Rate Codec (iLBC) and RTP Payload Format for iLBC” is supported for audio playback and for resending RTP streams.
- RFC 4145 – “TCP-Based Media Transport in the SDP” is partially supported.
- RFC 4235 – “An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP)” is supported.
- RFC 4244 – “An Extension to the Session Initiation Protocol (SIP) for Request History Information” is supported.
- RFC 4566, RFC 2327 – “SDP: Session Description Protocol” is supported, with the limitations and relaxations provided for SDP under SIP.
- RFC 4568 – “SDP Security Descriptions for Media Streams” is partially supported – for ordinary calls only.
- RFC 4572 – “Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the SDP)” is partially supported – for ordinary calls only.
- RFC 4961 – “Symmetric RTP/RTP Control Protocol (RTCP)” is supported, provided that PortaSIP is used to transport the media stream (the actual voice traffic) from one endpoint to another.
- RFC 5009 “Private Header (P-Header) Extension to the Session Initiation Protocol (SIP) for Authorization of Early Media” – supported. PortaSIP only re-sends the P-Early_Media SIP header.
- RFC 5502 – “The SIP P-Served-User Private-Header (P-Header) for the 3GPP IP Multimedia (IM) Core Network (CN) Subsystem” is partially supported. PortaSIP does not generate the P-Served-User header.
- RFC 5574 – “RTP Payload Format for the Speex Codec” is supported for audio playback and for resending RTP streams between end-points.
- RFC 5589 – “Session Initiation Protocol (SIP) Call Control – Transfer” is supported with the following limitation for semi-attended transfer:
- In the call flow described in the RFC, Music on Hold (MOH) is played for the user to whom the call is transferred to prevent them from hanging up prematurely. However, because this connection process is typically completed almost immediately in PortaSwitch, it doesn’t play MOH for the user to whom the call is transferred.
- RFC 5763 – “Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)” is partially supported – for ordinary calls only.
- RFC 5764 – “DTLS Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)” is partially supported – for ordinary calls only.
- RFC 5806 – “Diversion Indication in SIP” is supported.
- RFC 6035 – “Session Initiation Protocol Event Package for Voice Quality Reporting” is supported.
- RFC 6189 – “ZRTP: Media Path Key Agreement for Unicast Secure RTP” is partially supported – for ordinary calls only.
