Session Initiation Protocol (SIP)

Estudo sobre SIP, SIP-I, SIP-T e a comparação

• Atualmente, dois sistemas padrão estão disponíveis para a interoperabilidade entre softswitch SIP e a tradicional PSTN: o conjunto de protocolos SIP for Telephones (SIP-T), da IETF, e o conjunto de protocolos SIP with Encapsulated ISUP (SIP-I), da ITU-T.

SIP

• SIP é um conjunto de protocolos de sinalização para estabelecer, modificar e terminar sessões IP, incluindo a conferência multimídia sobre IP.

1. Algumas funções
   • Tradução de nome e localização de usuário: SIP utiliza um conjunto de mecanismos de nomes semelhantes para o endereço de e-mail. Cada usuário é determinado por todos os elementos ULR hierárquicos, por exemplo, SIP: usuário@company.com, 80000001@beijing, ou username@202.112.10.4. e pode ser associado com um endereço de E-mail diretamente.
   • Negociação das características de parâmetros de sessão: SIP permite que os participantes em uma chamada negociem as características dos parâmetros da sessão. Por exemplo, a função de videochamada não pode ser utilizada quando vários usuários de videochamada tem uma sessão com um usuário do telefone móvel. Depois de o usuário do telefone móvel encerrar a sessão, os usuários de videochamada podem renegociar sobre o uso a função de vídeo.
   • Gerenciamento de participantes na chamada: Durante uma sessão, os participantes podem convidar outros usuários para participar, transferir, manter e cancelar a conexão.
2. Componentes do SIP
   • SIP consiste de dois componentes básicos: agente de usuário SIP e servidor de rede SIP, onde as comunicações SIP são realizadas.Arquivo:SessionInitiationProtocol(SIP)ComponentesBasicos.jpg
   • The SIP user agent is actually the terminal system component of the call. The SIP network server is the network equipment that processes signaling related to multi-point call (IP conference), and it is responsible for resolving user names and locating users.
3. Applications of SIP
   • SIP supports powerful multimedia services and is extendable to support different applications. In fixed-line softswitch applications, SIP is located at the call control layer of the flat architecture, and supports call connection between different softswitches. From the viewpoint of routing, there are two cases for the usage of SIP architecture:
     • In the first case, normal ISDN User Protocol (ISUP) messages are encapsulated into SIP messages (with additional information added) for transmission. Functions such as call server, number, route analysis, signaling and service interworking remain unchanged, and route analysis routes the messages to the destination IP addresses.
     • The second case is based on the ENUM (telephone number mapping working group of IETF) database. In this case, the call control of the call server does not provide number and route analysis functions, which is absolutely different from that of current circuit-switched network. But service mapping and interworking are still required. Because the Circuit Identification Code (CIC), ISUP management process and Message Transfer Protocol (MTP) are not used, the standard ISUP needs to be modified correspondingly. #*Therefore, the network management is simplified (such as, there is no signaling network and routing definition). Also in this mode, the control of telecom carriers on networks is reduced and control modes have great changes, compared with current networks.
   • The aforesaid analysis shows that the application of SIP disables some functions of current telephone networks. To introduce these functions, you need to extend SIP.

SIP-T

• SIP-T is defined by RFC3372 of the IETFMMUSIC working group. The SIP-T protocol suite contains RFC3372, RFC2976, RFC3204 and RFC3398.
• SIP-T is an extension of SIP, which inherits the flexibility of SIP, provides additional support for telephone application and therefore is quite suitable for IP networks. SIP-T enables the SIP messages to carry the ISUP signaling. It established three models for interworking between SIP and ISUP on an end-to-end basis, that is, PSTN-PSTN call over an SIP network, SIP-PSTN call, and PSTN-SIP call.
• SIP-T provides two methods for the interworking between SIP and ISUP, that is, encapsulation and mapping, which are defined by RFC3204 and RFC3398 respectively. However, the SIP-T only focuses on the interworking of the basic call, and does not involve supplementary services basically.

SIP-I

• SIP-I contains TRQ.2815 and Q.1912.5 of the ITU-TSG11 working group. TRQ.2815 defines technical requirements for interworking between SIP and Bearer Independent Call Control Protocol (BICC)/ISUP, including interworking interface model, protocol capability set supported by InterWorking Unit (IWU), and security model of the interworking interface. Q.1912.5 defines the interworking between 3GPPSIP and BICC/ISUP, between SIP and BICC/ISUP and between SIP-I and BICC/ISUP in detail, according to different protocol capability sets supported by IWU at the Network-to Network Interface (NNI) on the SIP side.
• SIP-I reuses many IETF standards and drafts, which covers not only the interworking of the basic call, but the interworking of the BICC/ISUP supplementary service.

Comparison of SIP, SIP-T and SIP-I

• SIP can connect users that use any IP networks (wired LAN and WAN, public Internet backbone network, 2.5G, 3G and Wi-Fi mobile networks) and any IP equipment (telephone, PC, Personal Digital Assistant (PDA) and mobile handheld equipment), so it improves the communication mode between enterprise and user. Even if SIP-based applications (such as VOIP, multimedia conference, push-to-talk, location service, online information and IM) are used independently, they can provide many new business opportunities for service provider, Independent Software Vendor (ISV), and network equipment supplier and developer. However, the basic value of SIP lies in that it can combine these functions into seamless communication services to a larger degree.
• SIP-I and SIP-T are all extensions to SIP. However, SIP-I reuses many IETF standards and drafts and is richer than SIP-T in contents. SIP-I contains not only the interworking of the basic call, but also the interworking of supplementary services such as CLIP and CLIR. Besides interworking of the call signaling, SIP-I takes other issues into account such as resource reservation, media information conversion, and interworking between fixed-line SIP/3GPPSIP and BICC/ISUP. The most important is that SIP-I inherits advantages (such as clarity, preciseness and elaboration) of the traditional ITU-T recommendations and is much better than SIP-T in operability.
• At present, SIP is a dominant protocol that controls the communication between softswitch and application server, and will be the absolutely dominant protocol for communication between softswitch and terminal in the future. In the context of inter-softswitch communication, SIP - T has a wider application. For example, in CDMA2000 protocol, SIP-T is used between MSCes. SIP - I has been regarded as the core interworking protocol between softswitch and traditional telecommunication networks by 3GPP, main telecommunication carriers and big telecommunication equipment supplier around the world. (Li Zhaowei)

Referências:

http://wwwen.zte.com.cn/endata/magazine/ztetechnologies/2005year/no2/articles/200503/t20050321_161450.html

Pesquisador

André Carrijo de Oliveira