| RFC number | Title | Support | Notes |
|---|---|---|---|
| RFC 1889 | RTP: A Transport Protocol for Real-Time Applications | Partially supported | Limitation:
|
| RFC 2327 | SDP: Session Description Protocol | Supported | With the limitations and relaxations provided for SDP under SIP. |
| RFC 2543 | SIP: Session Initiation Protocol | Not supported | Obsolete features such as tag absence and shortened branch calculation are not supported and are known to confuse PortaSIP.
However, server-side transaction identification is compatible with RFC 2543 |
| RFC 2833 | RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals | Supported | |
| RFC 2976 | The SIP INFO Method | Partially supported | The PortaSIP cluster can either resend INFO requests to a remote UA or extract DTMF information if a call scenario requires it. |
| RFC 3261 | SIP: Session Initiation Protocol | Supported | Limitations:
|
| RFC 3262 | Reliability of Provisional Responses in the Session Initiation Protocol (SIP) | Fully supported | |
| RFC 3263 | Session Initiation Protocol (SIP): Locating SIP Servers | Partially supported | NAPTR records are not supported |
| RFC 3264 | An Offer/Answer Model with the Session Description Protocol (SDP) | Partially supported | For a late offer/answer model. |
| RFC 3265 | Session Initiation Protocol (SIP)-Specific Event Notification | Supported | |
| RFC 3311 | The Session Initiation Protocol (SIP) UPDATE Method | Supported | |
| RFC 3312 | Integration of Resource Management and Session Initiation Protocol (SIP) | Partially supported | B2BUA can resend the part of the SDP with preconditions or cut it before sending it to the opposite side. PortaSIP doesn’t generate preconditions or change the system behavior based on the precondition values. |
| RFC 3323 | A Privacy Mechanism for the Session Initiation Protocol (SIP) | Partially supported | |
| RFC 3324 | Short Term Requirements for Network Asserted Identity | Partially supported | |
| RFC 3325 | Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks | Partially supported | |
| RFC 3326 | The Reason Header Field for the Session Initiation Protocol (SIP) | 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 | Supported | |
| RFC 3428 | Session Initiation Protocol (SIP) Extension for Instant Messaging | Supported | |
| RFC 3489 | STUN – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) | Supported | |
| RFC 3515 | The Session Initiation Protocol (SIP) REFER Method | Partially supported | For attended transfer, PortaSIP does not update the original CLI when sending a request to the transfer target. |
| RFC 3550
|
RTP: A Transport Protocol for Real-Time Applications” | Partially supported | Limitation:
|
| RFC 3551 | RTP Profile for Audio and Video Conferences with Minimal Control | Supported | Not all encodings are supported. |
| RFC 3555 | MIME Type Registration of RTP Payload Formats | Supported | |
| RFC 3581 | An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing | Supported | |
| RFC 3611 | RTP Control Protocol Extended Reports (RTCP XR) | Supported | |
| RFC 3666 | Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows | Partially supported | |
| RFC 3711 | The Secure Real-time Transport Protocol (SRTP) | Supported | PortaSIP passes encrypted packets between the phones and does not perform any encryption. |
| RFC 3842 | A Message Summary and Message Waiting Indication Event Package for the SIP | Supported | |
| RFC 3856 | A Presence Event Package for the Session Initiation Protocol (SIP) | Supported | |
| RFC 3891 | The Session Initiation Protocol (SIP) ‘Replaces’ Header | Supported | |
| RFC 3892 | The Session Initiation Protocol (SIP) Referred-By Mechanism | Supported | |
| RFC 3903 | SIP Extension for Event State Publication | Supported | |
| RFC 3951
RFC 3952 |
Internet Low Bit Rate Codec (iLBC) and RTP Payload Format for iLBC | Supported | For audio playback and for resending RTP streams. |
| RFC 3960 | Early Media and Ringing Tone Generation in the Session Initiation Protocol | Partially supported | |
| RFC 4028 | Session Timers in the Session Initiation Protocol (SIP) | Partially supported | |
| RFC 4145 | TCP-Based Media Transport in the SDP | Partially supported | |
| RFC 4235 | An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP) | Supported | |
| RFC 4244 | An Extension to the Session Initiation Protocol (SIP) for Request History Information | Supported | |
| RFC 4566 | SDP: Session Description Protocol | Supported | With the limitations and relaxations provided for SDP under SIP. |
| RFC 4568 | SDP Security Descriptions for Media Streams | Partially supported | For ordinary calls only. |
| RFC 4572 | Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the SDP) | Partially supported | For ordinary calls only. |
| RFC 4855
RFC 4856 |
MIME Type Registration of RTP Payload Formats | Supported | |
| RFC 4961 | Symmetric RTP/RTP Control Protocol (RTCP) | 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 5245 | Session Controller supports Interactive Connectivity Establishment (ICE) | Partially supported | "lite" implementation of ICE. |
| RFC 5389 | Session Traversal Utilities for NAT (STUN) | Supported | |
| RFC 5502 | The SIP P-Served-User Private-Header (P-Header) for the 3GPP IP Multimedia (IM) Core Network (CN) Subsystem | Partially supported | PortaSIP does not generate the P-Served-User header. |
| RFC 5574 | RTP Payload Format for the Speex Codec | Supported | For audio playback and for resending RTP streams between endpoints. |
| RFC 5589 | RTP Payload Format for the Speex Codec | Supported | Limitation for semi-attended transfer:
|
| RFC 5621 | Message Body Handling in SIP | Partially supported | Limitation of one depth level (no subparts of subparts). |
| RFC 5626 | Managing Client-Initiated Connections in the Session Initiation Protocol (SIP) | Partially supported | |
| RFC 5763 | Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS) | Partially supported | For ordinary calls only. |
| RFC 5764 | DTLS Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP) | Partially supported | For ordinary calls only. |
| RFC 5766 | Traversal Using Relays around NAT (TURN) | Supported | |
| RFC 5806 | Diversion Indication in SIP | Supported | |
| RFC 6035 | Session Initiation Protocol Event Package for Voice Quality Reporting | Supported | |
| RFC 6189 | ZRTP: Media Path Key Agreement for Unicast Secure RTP | Partially supported | For ordinary calls only. |
| RFC 6337 | Session Initiation Protocol (SIP) Usage of the Offer/Answer Model | Supported | Except for the Offer/Answer Model pattern 5. |
| RFC 6665 | SIP-Specific Event Notification (subscription mechanism) | Supported | |
| RFC 7587 | RTP Payload Format for the Opus Speech and Audio Codec | Supported | |
| SMPP v3.4 | The SMPP (Short Message Peer-to-Peer) protocol | Supported |
