In this section:
Supported GSMA Standards
Standard | Description | |
---|---|---|
GSMA IPX | IP-based interconnection for wireless networks |
Supported IETF Standards
Standard | Title/Description | |
---|---|---|
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 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 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 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 8068 | Session Initiation Protocol (SIP) Recording Call Flows | |
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 |
Supported ITU Standards
Standard | Description | |
---|---|---|
G.722.1 | Low-complexity coding at 24 and 32 kbit/s for hands-free operation in systems with low frame loss | |
Q.1912.5 | SIP Interworking | |
T.38 v.3 | Procedures for real-time Group 3 facsimile communication over IP networks |
Supported NENA Standards
Standard | Description | |
---|---|---|
08-001 v2 | Interim VoIP Architecture for Enhanced 9-1-1 Services (i2) |
Supported SIP Forum Standards
Standard | Description | |
---|---|---|
SIPConnect 1.0 | Direct IP peering between SIP-enabled IP PBXs and VoIP service provider networks | |
SIPConnect 1.1 | SIP-PBX / Service Provider Interoperability | |
SIPConnect 2.0 | Updated guidelines for SIP-PBX / Service Provider Interoperability |
Supported 3GPP Standards
Standard | Description | |
---|---|---|
3GPP TS 24.229 v11.5.0 | IP multimedia call control protocol based on SIP and SDP | |
3GPP TS 23.167 | IP Multimedia Subsystem (IMS) emergency sessions. | |
3GPP TS 26.114 | IP Multimedia Subsystem (IMS); Multimedia telephony; Media handling and interaction | |
3GPP TS 29.214 | Policy and Charging Control over Rx reference point | |
3GPP TS 32.260 | IP Multimedia Subsystem (IMS) charging | |
3GPP TS 32.299 | Diameter Charging Applications | |
3GPP TS 33.328 | IP Multimedia Subsystem (IMS) media plane security | |
3GPP TS 26.441 (definition) | The Enhanced Voice Services (EVS) speech codec |
The following public websites provide standards definition details.