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6. VoIP (Voice over Internet Protocol)The Internet is a large packet switching network connected throughout the world to transport packets between endpoints. An endpoint may be a computer, telephone, fax machine, storage device, radio, television or other device. The Internet and endpoints use a special set of rules called Internet Protocol (IP) to transport packets among endpoints. Many other networks now use IP including local area networks in enterprises, home networks and wireless networks. The packet switching network such as the Internet uses statistical multiplexing and the packet delivery time is un-deterministic when the network is congested. As more and more bandwidth is available in the Internet and the Internet is virtually free, transporting real-time traffics such as audio and video become attractive and transporting Voice over Internet Protocol (VoIP) without problems became reality. In 1996, the ITU ratified a protocol, H.323, "Packet-based multimedia communications systems". The original title of H.323 was "Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service". H.323 was originally created to provide a mechanism for transporting video conferencing applications over LAN. It has rapidly evolved to address the growing needs of VoIP networks. In 1996, IETF attendees circulated Session Initiation Protocol (SIP) that became a prevalent protocol today. SIP was specified in RFC 2543, which was approved in 1999 and RFC 2543 was replaced with RFC 3261 in June 2002. 6.1. H.323H.323 is an umbrella Recommendation from the ITU that defines the protocols to provide audio, visual and data communication sessions in point-to-point and multipoint conferences over a packet network. H.323 was initially created to provide a mechanism for transporting video conferencing applications over LAN in early 1990s. The original title of H.323 was "Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service". It was the first protocol that provided Voice over IP (VoIP). The ITU ratified H.323 version in 1996 and published series of new versions. Currently H.323 version 6 published in June 2006 is available.
H.323 references many other ITU protocols as listed below. H.323 Related Standards ITU Recommendations: General Aspects H.323: Packet-based multimedia communications systems. This Recommendation describes H.323 entities and their roles. Signaling H.225.0: Call signaling protocols and media stream packetization for packet-based multimedia communication systems. This Recommendation describes the means by which audio, video, data, and control are associated, coded, and packetized for transport between H.323 entities. Security H.235.0: H.323 security: Framework for security in H series (H.323 and other H.245-based) multimedia systems. This Recommendation provides the security profile between H.323entities including terminal-to-gatekeeper, gatekeeper-to-gatekeeper, and H.323 gateway-to-gatekeeper. ITU Rec. H.235 (2003) content was reorganized into H.235.0 to .9 when revised in 2005. The map can be found in Appendices IV, V, and VI of H.235.0. Control H.245: Control protocol for multimedia communication. This Recommendation specifies syntax and semantics of terminal information messages and procedures to use them for in-band negotiation at the start of or during communication. Interworking H.246: Interworking of H-series multimedia terminals with H-series multimedia terminals and voice/voiceband terminals on GSTN and ISDN Directory and Supplementary Services H.350.1: Directory services architecture for H.323 H.450.1: Generic functional protocol for the support of supplementary services in H.323. H.450x series describes supplementary services including call transfer, call diversion, call hold, call park, call waiting, etc. Mobility H.510: Mobility of H.323 multimedia systems. This Recommendation deals with mobility aspects for H.323 systems above the transport layer. The main focus is on the support of terminal mobility. It does not cover handover procedures where active calls can be maintained during location changes. H.530: Symmetric security procedures for H.323 mobility in H.510. This Recommendation provides security procedures in H.323 mobility environments in H.510. 6.2. SIP (Session Initiation Protocol)Session Initiation Protocol (SIP) is a signaling protocol used for establishing, modifying and terminating sessions in an IP network. A session may be a simple two-way telephone call or a collaborative multi-media conference. SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A UA can function in User Agent Client (UAC) or a User Agent Server (UAS). A UAC initiates the SIP request and a UAS contacts the user when a SIP request is received and that returns a response on behalf of the user. The role of UAC or UAS lasts only for the duration of a transaction. In other words, if a piece of software responds to a request, it acts as a UAS for the duration of that transaction. If it generates a request later, it assumes the role of a UAC for the processing of that transaction. Typically, a SIP end point is capable of functioning as both a UAC and a UAS. Figure 6-2 illustrates the elements in a SIP network. User agents are SIP endpoints such as VoIP telephone. SIP uses servers to receive requests and send responses to those requests. Typical servers are proxies, redirect servers, and registrars. A proxy server helps route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. A redirect server generates re-direction responses to requests it receives, directing the client to contact an alternate set of URIs. A registrar accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles.
SIP needs other protocols to provide communications services including the Real-time Transport Protocol (RTP) (RFC 1889) for transporting real-time data and the Real-Time Streaming Protocol (RTSP) (RFC 2326) for controlling delivery of streaming media, the Media Gateway Control Protocol (MEGACO) (RFC 3525) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Description Protocol (SDP) (RFC 4566) for describing multimedia sessions. The IETF SIP Working Group is chartered to define SIP specifications and its family of extensions. Currently RFC 3261 specifies the SIP specifications. The IETF Session Initiation Protocol Project INvestiGation (SIPPING) working group is chartered to document the use of SIP for several applications related to telephony and multimedia, and to develop requirements for extensions to SIP needed for those applications. SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture and PacketCable PCS. SIP is also used for presence. SIP Related Standards IETF RFCs: General Aspects and Signaling RFC 3261: Session Initiation Protocol (SIP). This document describes Session Initiation Protocol (SIP), an application-layer signaling protocol for creating, modifying, and terminating sessions between participants. RFC 3263: SIP: Locating SIP Servers. This document describes DNS procedures to allow a client to resolve a SIP Uniform Resource Identifier (URI) into the IP address, port, and transport protocol of the next hop to contact Extensions RFC 2976: The SIP INFO Method. This document proposes the INFO method extension to the SIP, which allow for the carrying of session related control information (e.g., ISUP and ISDN signaling messages) that is generated during a session. RFC 3262: Reliability of Provisional Responses in the Session Initiation Protocol (SIP). This document specifies an extension to the SIP that provides reliable provisional response messages by using the option tag 100rel and the Provisional Response ACKnowledgement (PRACK) method. RFC 3265: SIP-Specific Event Notification. This document describes the Event Notification extension to the SIP, which provides an extensible framework by which SIP nodes can request notification from remote nodes indicating that certain events have occurred. RFC 3311: The Session Initiation Protocol UPDATE Method. This specification defines the UPDATE method for the SIP, which allows a client to update parameters of a session such as the set of media streams. RFC 3326: The Reason Header Field for the Session Initiation Protocol (SIP). The same SIP request can be issued for a variety of reasons. For example, a SIP CANCEL request can be issued if the call has completed on another branch or was abandoned before answer. This document defines a header field, Reason, that provides this information. RFC 3515: The Session Initiation Protocol (SIP) Refer Method. This document specifies an extension informing that the recipient REFER to a resource provided in the request. RFC 3428: Session Initiation Protocol (SIP) Extension for Instant Messaging. This document proposes the MESSAGE method, an extension to the SIP that allows the transfer of Instant Messages RFC 3840: Indicating User Agent Capabilities in the Session Initiation Protocol (SIP). This specification defines mechanisms by which a SIP user agent can convey its capabilities and characteristics to other user agents and to the registrar for its domain. This information is conveyed as parameters of the Contact header field. RFC 3841: Caller Preferences for the Session Initiation Protocol (SIP). This document describes a set of extensions to the SIP which allow a caller to express preferences about request handling in servers. RFC 3891: The Session Initiation Protocol (SIP) "Replaces" Header. This document defines a new header for use with SIP multi-party applications and call control. The Replaces header is used to logically replace an existing SIP dialog with a new SIP dialog. RFC 3892: The Session Initiation Protocol (SIP) Referred-By Mechanism. This document extends the REFER method, allowing the referrer to provide information about the REFER request to the refer target using the referee as an intermediary. RFC 3903: Session Initiation Protocol (SIP) Extension for Event State Publication. This document describes an extension to the SIP for publishing event state used within the SIP Events framework. RFC 3911: The Session Initiation Protocol (SIP) "Join" Header. This document defines a new header for use with SIP multi-party applications and call control. The Join header is used to logically join an existing SIP dialog with a new SIP dialog. RFC 4028: Session Timers in the Session Initiation Protocol (SIP). This document defines an extension that allows for a periodic refresh of SIP sessions through a re-INVITE or UPDATE request. The refresh allows both user agents and proxies to determine whether the SIP session is still active. RFC 4244: An Extension to the Session Initiation Protocol (SIP) for Request History Information. This document defines a standard mechanism for capturing the history information associated with a SIP request. This capability enables many enhanced services by providing the information as to how and why a call arrives at a specific application or user. Security and Privacy RFC 3323: A Privacy Mechanism for the Session Initiation Protocol (SIP). This document provides privacy (i.e., withholding of the identity of a person) requirements and mechanisms for the SIP. RFC 3329: Security Mechanism Agreement for the Session Initiation Protocol (SIP) Sessions. This document defines new functionality for negotiating the security mechanisms used between a SIP user agent and its next-hop SIP entity. This new functionality supplements the existing methods of choosing security mechanisms between SIP entities. Management RFC 4780: Management Information Base for the Session Initiation Protocol (SIP). SIP Telephone RFC 3372: Session Initiation Protocol for Telephones (SIP-T): Context and Architectures. This document taxonomizes the uses of PSTN-SIP gateways, provides uses cases, and identifies mechanisms necessary for interworking. The mechanisms detail how SIP provides for both 'encapsulation' (bridging the PSTN signaling across a SIP network) and 'translation' at gateways. Other RFC 3361: Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol (SIP) Servers. This document specifies a DHCP option that allows SIP clients to locate a local SIP server that is to be used for all outbound SIP requests, a so-called outbound proxy server. RFC 3486: Compressing the Session Initiation Protocol (SIP). This document describes a mechanism to signal that compression is desired for one or more SIP messages.
Other Protocols Used by SIP RFC 1889: RTP: A Transport Protocol for Real-Time Applications. The RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RFC 2326: Real Time Streaming Protocol (RTSP). The RTSP is an application-level protocol for control over the delivery of data with real-time properties. RFC 3525: Gateway Control Protocol Version 1. This document defines the protocol used between elements of a physically decomposed multimedia gateway, i.e., a Media Gateway and a Media Gateway Controller. RFC 3761: The E.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM). This document discusses the use of the Domain Name System (DNS) for storage of E.164 numbers. More specifically, how DNS can be used for identifying available services connected to one E.164 number. RFC 4566: Session Description Protocol (SDP). The document describes a protocol for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation.
SIP Related Work Groups (WGs) in IETF: bliss (Basic Level of Interoperability for SIP Services): This WG is to facilitate effective feature interoperability for features sharing common functional primitives utilizing SIP in heterogeneous network environments. Initial work will be focused on four functional primitives - Line sharing, parking, automatic handling, call queuing. No RFC has been published. No RFCs. p2psip (Peer-to-Peer Session Initiation Protocol The Peer-to-Peer): This WG is chartered to develop protocols and mechanisms for the use of the Session Initiation Protocol (SIP) in settings where the service of establishing and managing sessions is principally handled by a collection of intelligent endpoints, rather than centralized servers as in SIP as currently deployed. No RFC has been published. No RFCs. simple (SIP for Instant Messaging and Presence Leveraging Extensions: This WG focuses on the application of the SIP to the suite of services collectively known as instant messaging and presence (IMP). RFC 3856: A Presence Event Package for the Session Initiation Protocol (SIP). This document describes the usage of the SIP for subscriptions and notifications of presence. Presence, also known as presence information, conveys the ability and willingness of a user to communicate across a set of devices. RFC 2778 defines a model and terminology for describing systems that provide presence information. RFC 3857: A Watcher Information Event Template-Package for the Session Initiation Protocol (SIP). This document defines the watcher information template-package for the SIP event framework. Watcher information refers to the set of users subscribed to a particular resource within a particular event package. A user can subscribe to this information, and therefore learn about changes to it. sipping (Session Initiation Proposal Investigation The Session Initiation Protocol Project InvestiGation): This working group is chartered to document the use of SIP for several applications related to telephony and multimedia, and to develop requirements for extensions to SIP needed for those applications. RFC 3824: Using E.164 numbers with the Session Initiation Protocol (SIP). This document illustrates how the SIP and ENUM might work in concert, and clarifies the authoring and processing of ENUM records for SIP applications. RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping. This document describes a way to perform the mapping between two signaling protocols: the SIP and the Integrated Services Digital Network (ISDN) User Part (ISUP) of Signaling System No. 7 (SS7). RFC 3578: Mapping of of Integrated Services Digital Network (ISUP) Overlap Signaling to the Session Initiation Protocol (SIP). This document describes a way to map Integrated Services Digital Network User Part (ISUP) overlap signaling to SIP. RFC 3666: Session Initiation Protocol PSTN Call Flows. This document contains best current practice examples of SIP call flows showing interworking with the Public Switched Telephone Network (PSTN). Elements in these call flows include SIP User Agents, SIP Proxy Servers, and PSTN Gateways. RFC 4353: A Framework for Conferencing with the Session Initiation Protocol (SIP). This document defines a framework for how SIP conferencing can occur. This framework describes the overall architecture, terminology, and protocol components needed for multi-party conferencing. RFC 4575: A Session Initiation Protocol (SIP) Event Package for Conference State. This document defines a conference event package for tightly coupled conferences using the SIP events framework, along with a data format used in notifications for this package. RFC 4579: Session Initiation Protocol (SIP) Call Control - Conferencing for User Agents. This specification defines conferencing call control features for the SIP. This document builds on the Conferencing Requirements and Framework documents to define how a tightly coupled SIP conference works.
PacketCable Specifications: PacketCable is a CableLabs-led initiative that is aimed at developing interface specifications for delivering real-time multimedia services over two-way cable networks. PacketCable networks use SIP to enable multimedia services including IP telephony and multimedia conferencing. PKT-SP-RST-ACCT-I01-060927: PacketCable Residential SIP Telephony Accounting Specification. This specification defines the collection of usage data needed to support Accounting of Residential SIP Telephony (RST) Features. PKT-SP-RST-E-DVA-I01-060927: PacketCable Residential SIP Telephony E-DVA Specification. This specification defines the embedded Digital Voice Adaptor (E-DVA) requirements for the analog interface and for powering of the E-DVA. An embedded DVA is a DOCSIS cable modem (CM) integrated with a PacketCable DVA. PKT-SP-RSTF-I01-060927: PacketCable Residential SIP Telephony Feature Specification. This document specifies an implementation of common residential telephony features in a PacketCable network with SIP based User Equipment (UEs).> PKT-SP-RST-PACM-I01-0609: PacketCable Residential SIP Telephony Provisioning, Activation, Configuration and Management Specification. The document specifies an implementation of common residential telephony features in a PacketCable network with PacketCable User Equipment (UE). 6.3. MGCP (Media Gateway Control Protocol)MGCP is a master/slave protocol for decomposed multimedia gateway consisting a Call Agent (CA) or Media Gateway Controller (MGC) and a Media Gateway (MG). The MGC contains the call control intelligence and the MG is an endpoint, which includes the media functions such as conversion of signals between TDM voice and VoIP. The MGC instructs the MG to create, modify and delete connections in order to establish and control media sessions with other multimedia endpoints. Also, the MGC can instruct the endpoints to detect certain events and generate signals. The endpoints automatically communicate changes in service state to the MGC. Furthermore, the MGC can audit endpoints as well as the connections on endpoints. Figure 6-3 illustrates telephony gateways based on MGCP, where MGC controls MG and communicates with Signaling Gateway via SIGTRAN. MGCP may be used to deploy VoIP solutions. PacketCable uses MGCP for deploying VoIP services in cable network.
MGCP Related Standards IETF RFCs: RFC 3435 Media Gateway Control Protocol (MGCP) Version 1.0. This document defines an application programming interface and a corresponding protocol used between elements of a decomposed multimedia gateway. The decomposed multimedia gateway consists of a Call Agent, which contains the call control "intelligence", and a media gateway which contains the media functions, e.g., conversion from TDM voice to Voice over IP. RFC 2805: Media Gateway Control Protocol Architecture and Requirements. This document describes protocol requirements for the MGCP between a Media Gateway Controller and a Media Gateway. RFC 2897: Proposal for an MGCP Advanced Audio Package. This document describes a proposal to add a new event/signal package to the MGCP to control an ARF (Audio Resource Function), which may reside on a Media Gateway or specialized Audio Server. RFC 3064: MGCP CAS Packages. This document contains a collection of media gateway Channel Associated Signaling (CAS) packages for R1 CAS, North American CAS, CAS PBX interconnect as well as basic FXO support. RFC 3149: MGCP Business Phone Packages. RFC 3441: Asynchronous Transfer Mode (ATM) Package for the Media Gateway Control Protocol (MGCP). This document describes an ATM package for the MGCP. This package includes new Local Connection Options, ATM-specific events and signals, ATM connection parameters, a description of codec, and profile negotiation. RFC 3624: The Media Gateway Control Protocol (MGCP) Bulk Audit Package. This document describes the audit commands that only allow a Call Agent to audit endpoint and/or connection state one endpoint at a time. RFC 3660: Basic Media Gateway Control Protocol (MGCP) Packages. This document provides a basic set of MGCP packages including generic, line, trunk, handset, RTP, DTMF (Dual Tone Multifrequency), announcement server and script packages. RFC 3661: Media Gateway Control Protocol (MGCP) Return Code Usage RFC 3991: Media Gateway Control Protocol (MGCP) Redirect and Reset Package RFC 3992: Media Gateway Control Protocol (MGCP) Lockstep State Reporting Mechanism
MGCP Package Sub-registry: Packages can be registered with the IANA according to the procedure defined in section C.1 of RFC 3435. MGCP LocalConnectionOptions Sub-registry: LocalConnectionOptions can be registered with the IANA according to the procedure defined in section C.3 of RFC 3435.
PacketCable Specifications: PKT-SP-EC-MGCP-I11-050812: Network-Based Call Signaling Protocol Specification (NCS). This specification describes a profile of the MGCP for PacketCable embedded clients, which we will refer to as the PacketCable Network-based Call Signaling (NCS) protocol. PKT-SP-TGCP-I10-050812: PSTN Gateway Call Signaling Protocol Specification (TGCP). This document describes a PacketCable profile of an application programming interface called a Media Gateway Control Interface (MGCI) and MGCP for controlling VoIP PSTN Gateways from external call control elements. PKT-SP-NCS1.5-I03-070412: Network-Based Call Signaling Protocol. This document describes the NCS profile of an application programming interface (MGCI) and MGCP for controlling embedded clients (e.g., network elements that provides analog access or video lines to a VoIP network) from external call control elements. PKT-SP-TGCP1.5-I03-070412: PSTN Gateway Call Signaling Protocol Specification. This document describes a PacketCable¢rofile of an application programming interface called a Media Gateway Control Interface (MGCI) and MGCP for controlling VoIP PSTN Gateways from external call control elements.
ITU Recommendations: IPCalbecom is a project under ITU, which defines time-critical interactive services over cable television network using IP-protocol, in particular Voice and Video over IP. The following list MGCP related recommendations in IPCablecom. J.160: Architectural framework for the delivery of time-critical services over cable television networks using cable modems. This document describes all major system components and framework network interfaces needed for delivery of IPCablecom services. J.162: Network call signaling protocol for the delivery of time-critical services over cable television networks using cable modems. This document defines a profile of the Media Gateway Control Protocol, referred to as the Network-base Call Signaling (NCS) protocol, that is used for call signaling to embedded clients in the centralized call control architecture of IPCablecom. J.165: IPCablecom Internet signaling transport protocol (ISTP). This recommendation addresses the protocol to implement ITU SS7 used for signaling interconnection in a distributed IPCablecom PSTN Gateway architecture. Specifically, it defines themessages and procedures for transporting SS7 ISUP, and TCAP messages as defined by ITU specifications between the IPCablecom control functions (Media Gateway Controller and CallManagement Server) and the SS7 Signaling Gateway. J.166: IPCablecom Management Information Base (MIB) framework. J.171.0: IPCablecom trunking gateway control protocol (TGCP) J.171.1: IPCablecom trunking gateway control protocol (TGCP) Profile J.171.2: IPCablecom trunking gateway control protocol (TGCP) Profile 2 J.175: Audio server protocol. This Recommendation describes the architecture and protocols that are required for playing announcements in IPCablecom networks.
ANSI/SCTE Specifications: The following is technical specifications of the Society of Cable Telecommunications Engineers (SCTE) related to MGCP in supporting the cable telecommunications industry. ANSI/SCTE 24-3 2006: IPCablecom Part 3: Network Call Signaling Protocol for the Delivery of Time-Critical Services over Cable Television Using Data Modems ANSI/SCTE 24-8 2006: IPCablecom Part 8: Signaling Management Information Base (MIB) Requirements ANSI/SCTE 24-11 2006: IPCablecom Part 11: Internet Signaling Transport Protocol (ISTP) ANSI/SCTE 24-12 2006: IPCablecom Part 12: Trunking Gateway Control Protocol (TGCP) ANSI/SCTE 24-17 2002: (formerly DSS 02-14) IPCablecom Audio Server Protocol
PacketCable Specifications: PKT-SP-EC-MGCP-I11-050812: PacketCable Network-Based Call Signaling Protocol Specification. This specification describes a profile of the Media Gateway Control Protocol (MGCP) for PacketCable embedded clients, which we will refer to as the PacketCable Network-based Call Signaling (NCS) protocol. 6.4. Megaco/H.248Media Gateway Control (Megaco)/H.248 addresses the relationship between the Media Gateway (MG) and the Media Gateway Controller (MGC). The MG converts media provided in one type of network to the format required in another type of network. For example, a MG could terminate bearer channels from a switched circuit network (e.g., DS0s) and media streams from a packet network (e.g., RTP streams in an IP network). The MGC controls the parts of the call state that pertain to connection control for media channels in a MG.
Megaco/H.248 is essentially quite similar to MGCP from an architectural standpoint and the controller-to-gateway relationship. They are successors of the IPDC (IP Device Control Protocol) and the SGCP (Simple Gateway Control Protocol) protocol specifications as shown in Figure 6-4. Since MGCP has been the immediate predecessor of MEGACO, many functions in MGCP have found there way into the Megaco/H.248 specification. MEGACO is feature-rich and provides a better option to manufacturers to build products with advanced features. Megaco was jointly developed by the IETF and the ITU. H.248 is the ITU name for Megaco. Similar to MGCO, Megaco may be used to deploy VoIP services. Megaco/H.248 Related Standards IETF RFCs: RFC 3525: Gateway Control Protocol Version 1. This document defines the protocol used between elements of a physically decomposed multimedia gateway, i.e., a Media Gateway and a Media Gateway Controller. RFC 3054:
Megaco IP Phone Media Gateway Application Profile.
This
document specifies a particular application of the Megaco/H.248 Protocol for
control of Internet telephones and similar appliances: the Megaco IP Phone
Media Gateway. The telephone
itself is a Media Gateway (MG), controlled by the Megaco/H.248 Protocol,
with application control intelligence located in the Media
Gateway Controller (MGC). ITU Recommendations: H.248.1: Gateway control protocol: Version 3. This Recommendation defines the protocols used between elements of a physically decomposed multimedia gateway. H.248.2 to H.248.47 define packages that extend the applicability of the H.248.1 Gateway Control Protocol Recommendation. 6.5. SIGTRAN and SPIRITSSIGTRAN is the name of an IETF Working Group that address the transport of packet-based PSTN signaling over IP Networks. For interworking with PSTN, IP networks will need to transport signaling such as Q.931 or SS7 ISUP messages between IP nodes such as a Signaling Gateway and Media Gateway Controller or Media Gateway. Examples of SIGTRAN for transport include: - Transport of signaling between a Signaling Gateway and Media Gateway or Media Gateway Controller - Transport of signaling ("backhaul") from a Media Gateway to a Media Gateway Controller - Transport of TCAP between a Signaling Gateway and other IP nodes SIGTRAN applications include: - Internet dial-up remote access - IP telephony interworking with PSTN - Other services as identified IETF SPIRITS Working Group addresses how services supported by IP network entities can be started from IN (Intelligent Network) requests, as well as the protocol arrangements through which PSTN (Public Switched Telephone Network) can request actions to be carried out in the IP network in response to events occurring within the PSTN/IN. Figure 6-5 illustrates protocols used for interworking between the PSTN and the IP network.
SIGTRAN and SPIRITS Related Standards IETF SIGTRAN RFCs: RFC 2719: Framework Architecture for Signaling Transport. This document defines an architecture framework for transport of message-based signaling protocols over IP networks. The framework describes relationships between functional and physical entities exchanging signaling information, such as Signaling Gateways and Media Gateway Controllers. It identifies interfaces where signaling transport may be used and the functional and performance requirements that apply from existing Switched Circuit Network (SCN) signaling protocols. The following RFCs cover methods of encapsulating different signaling protocols. RFC 2960: Stream Control Transmission Protocol. This document describes the SCTP. SCTP is designed to transport PSTN signaling messages over IP networks, but is capable of broader applications. RFC 3331: Signaling System 7 (SS7) Message Transfer Part 2 (MTP2) - User Adaptation Layer RFC 3332: Signaling System 7 (SS7) Message Transfer Part 3 (MTP3) - User Adaptation Layer (M3UA) RFC 3868: Signaling Connection Control Part User Adaptation Layer (SUA) RFC 4165: Signaling System 7 (SS7) Message Transfer Part 2 (MTP2) - User Peer-to-Peer Adaptation Layer (M2PA) RFC 4129: Digital Private Network Signaling System (DPNSS)/Digital Access Signaling System 2 (DASS 2) Extensions to the IUA protocol RFC 4233: Integrated Services Digital Network (ISDN) Q.921-User Adaptation Layer RFC 4166: Telephony Signaling Transport over Stream Control Transmission Protocol (SCTP) Applicability RFC 4666: Signaling System 7 (SS7) Message Transfer Part 3 (MTP3) - User Adaptation Layer (M3UA)
IETF SPIRITS RFCs: RFC 3136: The SPIRITS Architecture. This document describes the architecture for supporting SPIRITS services, which are those originating in the PSTN and necessitating the interactions between the PSTN and the Internet. (e.g., Internet Call Waiting, Internet Caller-ID Delivery, and Internet Call Forwarding.) RFC 3910: The SPIRITS (Services in PSTN requesting Internet services) Protocol |
