ANSI ATIS 1000028-2008 IP Device (SIP UA) to Network Interface Standard.pdf

1、 AMERICAN NATIONAL STANDARD FOR TELECOMMUNICATIONS ATIS-1000028.2008(R2013) IP Device (SIP UA) to Network Interface Standard As a leading technology and solutions development organization, ATIS brings together the top global ICT companies to advance the industrys most-pressing business priorities. T

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10、ute require that action be taken periodically to reaffirm, revise, or withdraw this standard. Purchasers of American National Standards may receive current information on all standards by calling or writing the American National Standards Institute. Notice of Disclaimer Adaptive Multi-Rate (AMR); Sp

11、eech processing functions; General description (GSM 06.71 version 7.0.2 Release 1998) 2.2.4 Other5TIA-127 TIA-127-A, Enhanced Variable Rate Codec (EVRC) Speech Option 3 for Wideband Spread Spectrum Digital Systems (May 2004) TIA-1016 TIA-1016-A, Source-Controlled Variable-Rate Multimode Wideband Spe

12、ech Codec (VMR-WB), Service Options 62 and 63 for Spread Spectrum Systems (January 2006) 3 DEFINITIONS 3.1 Definitions In this document, the label “must” indicates that it must be conformant to all mandatory provisions of the corresponding specification. However, it should be noted that being “confo

13、rmant to all mandatory provisions” means that it is mandatory to be able to respond to all required messages/headers, but it may not be mandatory in all circumstances to be able to send certain messages/headers. In addition, the label “must” means that the functionality must be implemented, although

14、 it is recognized that carriers, enterprise networks and devices may choose to not activate the functionality in some or all circumstances. The label “must1” indicates that all mandatory provisions of the corresponding specification must be implemented, but configuration or negotiation determines wh

15、ether the feature is used. The label “shall” indicates a mandatory requirement of this American National Standard. The label “should” indicates that it should be conformant to the corresponding specification. The label “should2” is used in the case of draft specifications for which the requirement i

16、s likely to become a “must” in a future version of this profile, after the draft specification has been promoted to an RFC. However, because draft specifications are subject to change, it is not feasible to make such specifications a “must” for the purposes of this version of the profile. Similarly,

17、 the label “should 1,2” indicates the requirement may become a “must1” in a future version of this profile. The word may denotes a optional capability that could augment the standard. The standard is fully functional without the incorporation of this optional capability. The words “see text” indicat

18、es that a description of the conformance criteria follows in this text. “N/A” indicates the corresponding specification is not required. The following definitions are applicable to the SIP UNI framework as defined in this document: Device: A device is the instrument through which an end user is conn

19、ecting to the Carrier network and contains a User Agent and a user interface. A device is a specific instance of the more generic EUF. 4This document is available from the European Telecommunications Standards Institute (ETSI). 5This document is available from the Telecommunications Industry Associa

20、tion (TIA). ATIS-1000028.2008 8 User Agent: A User Agent (UA) is a SIP construct that represents the device in a SIP-based network. It has an IP network connection on one side and a user interface connection on the other side. User interface: the user interface is the means by which a user of the DE

21、VICE interacts with the User Agent in order to initiate and receive voice communication, and is presented through something like a telephone or PC. Carrier network: A Carrier network is a SIP-based VoIP network that contains one or more elements, at least one of which is an application server (AS),

22、which is responsible for applying service logic to call requests or media streams. Application Server: An Application Server (AS) is an entity in the Carrier network that is responsible for applying service logic to call requests or media streams. Application servers implement services, in combinati

23、on with service logic in the User Agent. EUF: The end-user functions (EUF) includes end-user equipment, both the legacy terminals and NGN terminals, and also includes customer networks. End-user equipment may be either mobile or fixed. The end-user interfaces via which the EUF is connected to NGN ar

24、e supported by both physical and functional (control) interfaces. A SIP EUF contains a User Agent. SCF: The service control functions (SCF) establish, monitor, support, and release multimedia sessions and manage the users service interactions. (SIP) Proxy, Proxy Server: A Proxy or Proxy Server is an

25、 intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is sent to another entity “closer“ to the targeted user. Proxies are also us

26、eful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it. (SIP) B2B UA: A back-to-back user agent (B2BUA) is a concatenation of a SIP user agent client (UAC) and use

27、r agent server (UAS). Note: The IETF defines the B2BUA in RFC 3261 as “a logical entity that receives a request and processes it as a user agent server (UAS). In order to determine how the request should be answered, it acts as a user agent client (UAC) and generates requests. Unlike a proxy server,

28、 it maintains dialog state and shall participate in all requests sent on the dialogs it has established. Since it is a concatenation of a UAC and UAS, no explicit definitions are needed for its behavior.” (UAC and UAS behaviours are defined in RFC 3261.) A B2BUA reformulates a message before sending

29、 it as a new request. SIP PBX: A Private Branch eXchange is a telephone exchange that serves a particular business or office, as opposed to one a common carrier or telephone company operates for many businesses or for the general public. A SIP PBX has a SIP interface to the Carrier network. The SIP

30、PBX may implement a SIP Proxy or a B2BUA. The SIP PBX, in the context of this specification, appears to the network as if it is a device or EUF, even though the User Agent may be in the PBX, or in the physical device connected to the PBX. User Network Interface: The User-Network Interface (UNI) is t

31、he interface between the User Agent (UA) and the network. Services: The collection of behaviors to be applied to communications requests on behalf of the end user. 4 ABBREVIATIONS 4.1 Abbreviations ABNF Augmented BNF ATIS-1000028.2008 9 AMR Adaptive Multirate (codec) AMR NB AMR Narrowband AMR WB AMR

32、 Wideband ANSI American National Standards Institute B2BUA Back-to-Back User Agent BNF Backus-Naur Form CPE Customer Premises Equipment CSC-FE Call Session Control Functional Entity DNS Domain Name Service DTMF Dual Tone Multi Frequency EUF End-User Functions EVRC Enhanced Variable Rate Codec FQDN F

33、ully Qualified Domain Name HTTP Hypertext Transport Protocol IBC-FE Interconnection Border gateway Control Functional Entity IETF Internet Engineering Task Force IP Internet Protocol ISDN Integrated Services Digital Network ISO/IEC International Standardization Organization/International Electrotech

34、nical Commission ITU-T International Telecommunications Union Telecommunication IVR Interactive Voice Response KPML Key Press Markup Language MIME Multipurpose Internet Mail Extensions MPEG Moving Picture Experts Group MWI Message Waiting Indication NAT Network Address TranslationNGN Next Generation

35、 Network NGN-TE NGN Terminal Equipment PBX Private Branch eXchange PCM Pulse Code Modulation PoC Push to talk over Cellular PSTN Public Switched Telephone Network QoS Quality of Service RFC Request for Comments RTCP RTP Control Protocol RTP Real-time Transport Protocol SCF Service Control Functions

36、SDP Session Description Protocol SIP Session Initiation Protocol SIPS Session Initiation Protocol Secure SMV Selectable Mode Vocoders SRTP Secure Real-time Transport Protocol TCP Transmission Control Protocol TLS Transport Layer Security UA User Agent UAC User Agent Client UAS User Agent Server UDP

37、User Datagram Protocol UI User-Interface UNI User-Network Interface URI Uniform Resource Identifier VMR-WB Variable-Rate Multi-Mode Wideband VoIP Voice over IP VPN Virtual Private Network WPA Wi-Fi Protected Access XML Extensible Markup Language 5 REFERENCE MODEL The User-Network Interface (UNI) is

38、the interface between the User Agent (UA) and the network. The network is viewed here as more or less a single entity, although in fact different signaling will interact with different network elements. ATIS-1000028.2008 10 Figure 1.a: User-Network Interface Model for SIP phone connected directly to

39、 Carrier Network The UNI for SIP phones connected to the SIP network covers both residential and enterprise applications where the customer wants to take full advantage of the future SIP multi-media capabilities. This specification applies equally to SIP “Hard Phones” and “Soft Phones”. The services

40、 are implemented in the SIP UA, in the Application Server, or in a combination of the two, depending on the specific service. Service Provider SIP PBXFigure 1.b: User-Network Interface Model for SIP PBX to Service Provider The SIP PBX to Service provider scenario is aimed at the enterprise market. T

41、he UNI allows for the SIP PBX to be managed by either the enterprise, or the Service Provider. The UNI does not restrict the types of terminals, and can include both SIP “Hard Phones” and “Soft Phones”. In the case of SIP Phones, the SIP PBX can act as either a proxy or a B2B UA. This specification

42、assumes that the PBX will register users individually, however it is recognized that there may be value in supporting the capability for a SIP PBX to bulk register users. This later case, referred to as “aggregate end points”, is for further study. ATIS-1000028.2008 11 6 ASSUMPTIONS This collection

43、of baseline protocols is designed to enable implementation of basic function phones as well as rich feature phones. This informative section is intended to provide some background for the collection of normative specifications defined in the sections that follow. The specification in this document a

44、pplies to the implementation of the components on either side of the SIP UNI. The specification does not in general apply to the use of these components, which is a local matter for the carrier. For example, this document specifies that TLS shall be supported. This means that equipment must include

45、the functionality necessary to initiate calls using TLS. However, the carrier may choose not to use it for a given call, or even to disable this capability in a given deployment. This specification is intended to maximize interoperability, not to restrict a carriers business decisions. Using TLS ove

46、r TCP is the preferred transport for SIP messages, however it is recognized that many SIP devices use UDP, so UDP is also a requirement of the Network Edge. In addition, devices shall support SRTP for media encryption, although some configurations may not use SRTP for some calls. Other encryption is

47、 also possible but not required, such as IPSec as a VPN-tunneling mechanism to the service provider. IPSec is of particular interest for several reasons including: its common use over numerous interconnections, such as WiFi which may or may not also have additional security mechanisms such as WPA; a

48、nd its use in the PacketCable Security Specification, http:/ January 2005. Symmetric RTP and outbound proxy support is part of the Baseline Profile so that signaling and media can traverse typical consumer firewalls and NATs using “network hosted NAT traversal”, with minimum impact on the device imp

49、lementation. By adjusting the SIP Registration timeout to keep a NAT pinhole open, (for example, through correction of IP addresses and port numbers in SIP messages and SDPs) symmetric UDP allows the media to flow through typical firewalls and NATs, and can be directed to the appropriate target. There are deployment topologies in which symmetric RTP is not required. The outbound proxy for a device may be set. The setting may require that all signaling packets shall be sent to the outbound proxy or that only in