Comparison between BICC and SIP-NGN Protocol

BICC vs sip-Comparison of NGN Protocol

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One notable feature of the Next Generation Network Based on Softswitch is the separation of Call Control and bearer control. In terms of call and control of communication, from the Standard Research of the International Telecommunication Union (ITU-T) and Internet Engineering Task Group (IETF), there are two protocols that deserve attention, that is BICC (bearing independent Call Control Protocol) and SIP-T (sip for telephony ).

Reporting

(1) BICC

BICC was developed by the sg11 group of ITU-T to solve the problem of separation of Call Control and bearer control, so that call control signaling can be carried on various networks. For example, MTP (message transmission part) SS7 network, ATM network, and IP network. BICC evolved from ISUP (ISDN User segment) and is an important supporting tool for the evolution from traditional telecom networks to integrated multi-service networks.

Currently, BICC protocol is evolving from CS1 (capability set 1) to CS2 and cs3. CS1 supports the bearer of call control signals on MTP SS7 and ATM networks, and CS2 supports the bearer over IP networks, CS3 focuses on the quality of bearer applications such as MPLS and IP QoS and the intercommunication with SIP. Many equipment manufacturers and operators have participated in the formulation of CS3 standards.

(2) SIP-T

Before introducing the SIP-T, we need to understand the SIP protocol first. SIP is a Session Initiation Protocol. It is one of the multimedia communication system framework protocols developed by IETF. It is a text-based application-layer control protocol, independent of the underlying protocol, used to establish, modify, and terminate multi-party multimedia sessions on the IP network. The SIP protocol uses HTTP, SMTP, and other protocols to support proxy, redirection, user registration, and other functions, and supports user mobility. Through cooperation with RTP/RTCP (Real-Time Protocol/real-time control protocol), SDP (Media Description Protocol), RTSP (Real-Time stream protocol), and DNS (Domain Name Server, supports voice, video, Data, email, status, instant messaging, chat, etc.

SIP-T is an extension protocol of SIP, which increases support for telephone applications, inherits the flexibility of SIP, and is more suitable for IP networks. The extended SIP-T enables SIP messages to carry ISUP signaling, providing a mechanism for communication between SS7-based PSTN network users and SIP-based IP phone network users.

Comparison

In general, BICC is directly oriented to telephone applications. It comes from the traditional telecom camp and has a more rigorous system architecture, therefore, it can provide good transparency for implementing services in the existing circuit switching telephone network in NGN. In contrast, the SIP architecture is not as complete as BICC defines. Sip is mainly used to support multimedia and other new services, and has more flexible and convenient features in multiple service applications based on IP networks.

When the BICC architecture is adopted, all current functions can be kept unchanged, such as number and route analysis. The routing concept is still used. This means that the network management method is very similar to the existing circuit switching network.

And if the adoption of SIP-T system architecture, the situation is different. From the routing point of view, there are two possible ways to introduce SIP-T in NGN: one is to retain the concept of routing, that is, the routing has no meaning in the SIP environment; the other is to change the route to cater to the SIP environment.

In the first case, the functions of the call server, number, route analysis, and communication between services remain unchanged, and the routing analysis guides the addressing of the target IP address, A normal ISUP message is encapsulated in a SIP Message for transmission. In this case, the SIP-T can be seen as a new protocol that attaches encapsulated information to the ISUP. The second case is based on the enum (IETF's telephone number ing workgroup) database. In this way, the call control of the call server is completely different from the call control in the existing circuit exchange network. There will be no number and route analysis in Call Control, but service ing and intercommunication are still required. Because the circuit identification code CIC, ISUP management process, and Message Transfer Protocol MTP are not used, the standard ISUP protocol must be modified accordingly. The process for handling emergencies, such as the mismatch between SIP and ISUP messages, restart, and overload processing, must be sound. Network management is simplified to a certain extent (for example, no signaling network or route definition is required ). In addition, compared with the existing network, the operator's control over the network is reduced, and the control mode has undergone great changes.

Through the above analysis, we can see that using SIP-T protocols will lose some of the functionality of existing telephone networks to some extent. To introduce these features, you need to scale the SIP-T protocol. BICC provides basically all existing telephone network functions. I believe that the modified and standardized SIP-T can achieve BICC's support for traditional businesses, but it needs to be clear which network principle is used to apply which "routing" processing method.