Standard | Title/Description |
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RFC 2327 | SDP |
RFC 2617 | HTTP Authentication: Basic and Digest Access Authentication |
RFC 2671 | Extension Mechanisms for DNS (EDNS0) |
RFC 2833 | RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals |
RFC 2865 | Remote Authentication Dial In User Service (RADIUS) |
RFC 2915 | The Naming Authority Pointer (NAPTR) DNS Resource Record |
RFC 2976 | SIP INFO Method |
RFC 3087 | Control of Service Context using SIP Request-URI |
RFC 3204 | MIME media type ISUP |
RFC 3261 | Session Initiation Protocol |
RFC 3262 | Reliability of Provisional Responses |
RFC 3264 | An Offer/Answer Model with the Session Description Protocol (SDP) |
RFC 3265 | SIP-Specific Event Notification |
RFC 3267 (2002) | Real-time Transport Protocol RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs |
RFC 3311 | SIP UPDATE method |
RFC 3312 | Integration of Resource Management and SIP |
RFC 3313 | Private Session Initiation Protocol (SIP) Extensions for Media Authorization NOTE: Supported in pass-through mode only. |
RFC 3323 | A Privacy Mechanism for the Session Initiation Protocol (SIP) |
RFC 3324 | Short Term Requirements for Network Asserted Identity |
RFC 3325 | Asserted Identity |
RFC 3326 (2002) | The Reason Header Field for the Session Initiation Protocol (SIP) |
RFC 3327 | SP Extension Header Field for Registering Non-Adjacent Contacts |
RFC 3372 | SIP-I |
RFC 3389 | Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN) |
RFC 3398 | ISUP to SIP Mapping |
RFC 3411 | An Architecture for Describing SNMP Management Frameworks |
RFC 3412 | Message Processing and Dispatching for SNMP |
RFC 3413 | Simple Network Management Protocol (SNMP) Applications |
RFC 3414 | User-based Security Model (USM) for version 3 of the Simple Network Management Protocol (SNMPv3) |
RFC 3415 | View-based Access Control Model (VACM) for the SNMP |
RFC 3418 | Management Information Base (MIB) for the SNMP |
RFC 3420 | Internet Media Type message/sipfrag |
RFC 3428 | SIP Extension for Instant Messaging |
RFC 3515 | SIP Refer Method |
RFC 3550 (2003) | RTP: A Transport Protocol for Real-Time Applications |
RFC 3551 (2003) | RTP Profile for Audio and Video Conferences with Minimal Control |
RFC 3581 | SIP Extension for Symmetric Response Routing |
RFC 3584 | Coexistence between Version 1, 2 and 3 of the Internet-standard Network Management Framework |
RFC 3605 | RTCP Attribute in SDP |
RFC 3608 | Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration |
RFC 3680 | A Session Initiation Protocol (SIP) Event Package for Registrations |
RFC 3711 | Secure Real-time Transport Protocol |
RFC 3824 | Using E164 numbers with the Session Initiation Protocol (SIP) |
RFC 3826 | The Advanced Encryption Standard (AES) Cipher Algorithm in the SNMP User-based Security Model |
RFC 3840 | Indicating User Agent Capabilities in SIP |
RFC 3841 | Caller Preferences for the Session Initiation Protocol (SIP) |
RFC 3842 | A Message Summary and Message Waiting Indication Event Package for the SIP |
RFC 3891 | The Session Initiation Protocol (SIP) "Replaces" Header |
RFC 3892 | The SIP Referred-By Mechanism |
RFC 3960 | Early Media and Ringback Tone Generation in the Session Initiation Protocol |
RFC 3966 | The tel URI for Telephone Numbers |
RFC 4022 | Management Information Base for the Transmission Control Protocol (TCP) |
RFC 4028 | Session Timers in SIP |
RFC 4103 | RTP Payload for Text Conversation |
RFC 4113 | Management Information Base for the User Datagram Protocol (UDP) |
RFC 4117 | Transcoding Services Invocation in the Session Initiation Protocol (SIP) Using Third Party Call Control (3pcc) |
RFC 4123 | Session Initiation Protocol (SIP)-H.323 Interworking Requirements |
RFC 4145 | TCP-Based Media Transport in the Session Description Protocol (SDP) |
RFC 4244 | An Extension to the Session Initiation Protocol (SIP) for Request History Information |
RFC 4293 | Management Information Base for the Internet Protocol (IP) |
RFC 4412 | Communications Resource Priority for the Session Initiation Protocol (SIP) |
RFC 4458 | Session Initiation Protocol (SIP) URIs for Applications - Voicemail and Interactive Voice Response (IVR) |
RFC 4566 | Session Description Protocol |
RFC 4567 | Key Management Extensions for SDP and RTSP |
RFC 4568 | SDP Security Descriptions for Media Streams |
RFC 4572 | Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP) |
RFC 4573 | MIME Type Registration for RTP Payload Format for H.224 |
RFC 4583 | Session Description Protocol (SDP) Format for Binary Floor Control Protocol (BFCP) Streams |
RFC 4585 | Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF) |
RFC 4730 | A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus (KPML) |
RFC 4753 | ECP Groups for IKE and IKEv2 |
RFC 4861 | Neighbor Discovery for IP version 6 (IPv6) |
RFC 4961 | Symmetric RTP / RTP Control Protocol (RTCP) |
RFC 4975 | Message Session Relay Protocol (MSRP) |
RFC 4976 | Relay Extensions for the Message Session Relay Protocol (MSRP) |
RFC 5104 | Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF) |
RFC 5168 | XML Schema for Media Control |
RFC 5227 | IPv4 Address Conflict Detection |
RFC 5343 | Simple Network Management Protocol (SNMP) Context EngineID Discovery |
RFC 5503 | Private Session Initiation Protocol (SIP) Proxy-to-Proxy Extensions for Supporting the PacketCable Distributed Call Signaling Architecture Note: Partial support only for the P-DCS-LAES header. |
RFC 5547 | SDP Offer/Answer Mechanism To Enable File Transfer |
RFC 5590 | Transport Subsystem for the Simple Network Management Protocol (SNMP) |
RFC 5626 | Managing Client-Initiated Connections in SIP |
RFC 5627 | Obtaining and Using Globally Routable User Agent URIs (GRUUs) in the Session Initiation Protocol (SIP) |
RFC 5705 | Keying Material Exporters for Transport Layer Security (TLS) |
RFC 5763
| Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS) |
RFC 5764 | Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP) |
RFC 5806 | Diversion Indication in SIP |
RFC 5880 | Bidirectional Forwarding Detection (BFD) |
RFC 5881 | Bidirectional Forwarding Detection (BFD) for IPv4 and IPv6 (Single Hop) |
RFC 5883 | Bidirectional Forwarding Detection (BFD) for Multihop Paths |
RFC 5996 | Internet Key Exchange Protocol Version 2 (IKEv2) |
RFC 6044 | Mapping and Interworking of Diversion Information between Diversion and History-Info Headers in SIP |
RFC 6135 | Alternative Connection Model for MSRP UA |
RFC 6140 | RFC 6140—Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP) |
RFC 6223 | Indication of Support for Keep-Alive |
RFC 6337 | Session Initiation Protocol (SIP) Usage of the Offer/Answer Model |
RFC 6347 | Datagram Transport Layer Security Version 1.2 |
RFC 6442 | Location Conveyance for the Session Initiation Protocol |
RFC 6714 | Connection Establishment for Media Anchoring |
RFC 6716 | Definition of the Opus Audio Codec |
RFC 6733 | Diameter Base Protocol |
RFC 6891 | Extension Mechanisms for DNS (EDNS (0) |
RFC 6961 | The Transport Layer Security (TLS) Multiple Certificate Status Request Extension |
RFC 7135 | Registering a SIP Resource Priority Header Field Namespace for Local Emergency Communications |
RFC 7245 | An Architecture for Media Recording Using the Session Initiation Protocol |
RFC 7865 | Session Initiation Protocol (SIP) Recording Metadata
|
RFC 7866 | Session Recording Protocol |
RFC 6377 | Non-reliable provisional response to the INVITE |
RFC 3261 | Offer-Answer Timer |
RFC 8068 | Session Initiation Protocol (SIP) Recording Call Flows |
RFC 8443 | Personal Assertion Token (PASSporT) Extension for Resource Priority Authorization |
draft-yu-tel-url-08 | New Parameters for the 'tel' URL to Support Number Portability and Freephone Service |
draft-ietf-payload-rtp-opus-03 | RTP Payload Format for Opus Speech and Audio Codec |
draft-spittka-silk-payload-format-00 | SILK codec draft standard |