The table below shows whether a particular standard is supported in Rhino SIP Servlet itself ("Full"), or if the standard may be easily supported with application code ("Partial").
Most SIP standards require a basic SIP stack, and add extensions that can be implemented at the application layer. The rapid pace of SIP development means it is not feasible to implement every new standard directly in the container, and this is often not required when applications can easily be modified to support new standards.
| Specification | Description | Compliance | Notes |
|---|---|---|---|
The PINT Service Protocol: Extensions to SIP and SDP for IP Access to Telephone Call Services |
Partial |
||
INFO Method |
Full |
||
SIP: Session Initiation Protocol |
Full |
||
Reliability of Provisional Responses |
Full |
||
SIP: Locating SIP Servers |
Full |
||
SIP-Specific Event Notification |
Full |
||
UPDATE Method |
Full |
||
Integration of Resource Management and SIP |
Full |
||
Private SIP Extensions for Media Authorization |
Full |
||
Signalling Compression (SigComp) |
None |
Not normally needed on application servers |
|
A Privacy Mechanism for SIP |
Full |
||
Private Extensions to SIP for Asserted Identity within Trusted Networks |
Full |
||
The Reason Header Field for the Session Initiation Protocol |
Full |
||
SIP Extension Header Field for Registering Non-Adjacent Contacts |
Full |
||
Security Mechanism Agreement for SIP |
Partial |
||
SIP for Telephones (SIP-T) |
Partial |
Best current practice RFC |
|
ISUP to SIP Mapping |
Partial |
||
Extension for Instant Messaging |
Full |
||
P-Header Extensions to SIP for 3GPP |
Full |
||
The SIP Refer Method |
Full |
||
Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to SIP |
Partial |
||
An Extension to SIP for Symmetric Response Routing |
Full |
||
Private SIP Proxy-to-Proxy Extensions for Supporting the PacketCable Distributed Call Signaling Architecture |
Partial |
||
SIP Extension Header Field for Service Route Discovery During Registration: Service-Route header |
Full |
||
A SIP Event Package for Registrations |
Partial |
||
Authentication, Authorization, and Accounting Requirements for the Session Initiation Protocol |
N/A |
Informational RFC |
|
Best Current Practices for Third-Party Call Control |
N/A |
Best current practice RFC |
|
Enumservice Registration for SIP Addresses-of-Record |
Partial |
Application must implement ENUM DNS lookups. |
|
Indicating User Agent Capabilities in SIP |
Full |
||
Caller Preferences for SIP |
Partial |
||
A Message Summary and Message Waiting Indication Event Package for SIP |
Partial |
||
S/MIME Advanced Encryption Standard (AES) Requirement for SIP |
Partial |
||
A Presence Event Package for SIP |
Partial |
||
A Watcher Information Event Template-Package for SIP |
Partial |
||
The SIP "Replaces" Header |
Partial |
||
The SIP Referred-By Mechanism |
Partial |
||
SIP Authenticated Identity Body (AIB) Format |
Partial |
||
SIP Extension for Event State Publication |
Full |
Supports PUBLISH method |
|
Join header |
Partial |
||
The Early Session Disposition Type for SIP |
Partial |
||
The tel URI for Telephone Numbers |
Full |
||
Session Timers in the Session Initiation Protocol |
Partial |
Timers must be implemented by the application. May support session timers automatically in the future. |
|
Update to the SIP Preconditions Framework |
Full |
||
Usage of the Session Description Protocol (SDP) Alternative Network Address Types (ANAT) Semantics in SIP |
Partial |
||
The Stream Control Transmission Protocol (SCTP) as a Transport for the SIP |
None |
Rhino SIP Servlet supports TCP, UDP and TLS. SCTP support is planned. |
|
An INVITE-Initiated Dialog Event Package for SIP |
Partial |
||
An Extension to SIP for Request History Information |
Partial |
||
Actions Addressing Identified Issues with the SIP Non-INVITE Transaction |
Partial |
||
Extending the SIP Reason Header for Preemption Events |
Partial |
||
Communications Resource Priority for SIP |
Partial |
||
The SIP P-User-Database Private-Header |
Partial |
||
Enhancements for Authenticated Identity Management in SIP |
Partial |
||
A Mechanism for Content Indirection in SIP Messages |
Partial |
||
Suppression of SIP REFER Method Implicit Subscription |
Partial |
||
Conveying Feature Tags with the SIP REFER Method |
Full |
||
Request Authorization through Dialog Identification in SIP |
Partial |
||
A SIP Event Package for Conference State |
Partial |
||
A SIP Event Package for Key Press Stimulus (KPML) |
Partial |
||
Management Information Base for the SIP |
None |
||
Connected Identity for the SIP |
Partial |
||
The P-Answer-State Header Extension to SIP for the Open Mobile Alliance Push to Talk over Cellular |
Partial |
||
The SIP P-Profile-Key Private Header (P-Header) |
Partial |
||
Rejecting Anonymous Requests in the SIP |
Full |
||
Private Header (P-Header) Extension to the SIP for Authorization of Early Media: P-Early-Media |
Partial |
||
A Framework for Consent-Based Communications in the SIP |
Partial |
||
The SIP Pending Additions Event Package |
Partial |
||
Multiple-Recipient MESSAGE Requests in the SIP |
Full |
||
Conference Establishment Using Request-Contained Lists in the SIP |
Partial |
||
Subscriptions to Request-Contained Resource Lists in the SIP |
Partial |
||
Referring to Multiple Resources in the SIP |
Full |
||
Requesting Answering Modes for SIP |
Partial |
||
Addressing an Amplification Vulnerability in SIP Forking Proxies |
Partial |
||
The SIP P-Served-User P-Header for the 3GPP IM CN Subsystem |
Partial |
||
Message Body Handling in SIP |
Partial |
||
Managing Client-Initiated Connections in SIP |
Partial |
RA supports an earlier draft, draft-ietf-sip-outbound-03. |
|
Obtaining and Using Globally Routable User Agent URIs (GRUUs) in SIP |
Partial |
||
Registration Event Package Extension for SIP GRUUs |
Partial |
||
Use of the SIPS URI Scheme in SIP |
Partial |
||
Addressing Record-Route Issues in SIP |
Partial |
||
A SIP Media Feature Tag for MIME Application Subtypes |
Partial |
||
IP Multimedia (IM) session handling; IM call model; Stage 2 |
Full |
||
IP Multimedia Call Control Protocol based on SIP and Session Description Protocol (SDP) |
Partial |
Application must implement AS behaviour. |
|
Presence service based on SIP; Functional models, information flows and protocol details |
Partial |
||
Interworking between Session Initiation Protocol (SIP) and Bearer Independent Call Control protocol or ISDN User Part (SIP-I) |
Partial |
||
SIP Servlet API v1.1 |
Full |
Passes JSR289 TCK |
