BICC VS SIP - NGN Agreement
-------------------------------------------------- ------------------------------
A significant feature of the next generation network based on soft exchange is to be able to achieve call control and bearer control phase separation. In terms of communication call and control, from the standard research of the International Telecommunication Alliance (ITU-T) and Internet Engineering Task Force (IETF), there are two research and development of two agreements worthy of attention, that is, BICC (carrying independence) Call Control Protocol) and SIP-T (SIP for Telephony).
Various newspapers
(1) BICC
BICC is developed by the ITU-T SG11 group, which is the main purpose to solve the problem of call control and carrying control separation, so that the call control signaling can be loaded on a variety of networks. Such as MTP (Message Delivery section) SS7 network, ATM network, IP network, etc. BICC is evolved from the ISUP (ISDN user) and is an important support tool for traditional telecommunications networks to integrated multi-service network evolution.
At present, the BICC protocol is developed from CS1 (capable set 1) to CS2 and CS3. CS1 supports call control signaling on MTP SS7, ATM network bearer, CS2 increases the bearer on IP online, and CS3 focuses on MPLS, IP QOS, etc. carrying the quality of the application and interact with SIP. Many equipment manufacturers and operators are involved in the formulation of CS3 standards.
(2) SIP-T
Before introducing SIP-T, we must first understand the SIP protocol. SIP is a session initiating protocol, which is one of the Multimedia Communication System Framework Agreement established by IETF. It is a text-based application layer control protocol, independent of the underlying protocol, used to create, modify and terminate both parties or multimedia sessions on IP online. The SIP protocol draws on an agreement such as HTTP, SMTP, supports agents, redirects, and registers locates such as users, and supports user movement. SIP supports voice, videos, data, email, status,, support voice, video, data, email, status, and Timely message, chat, etc.
SIP-T is the SIP expansion protocol, increasing support for telephone applications, inheriting the flexibility of SIP, and is more suitable for IP networks. The extended SIP-T allows SIP messages to provide interworking mechanisms for SS7-based PSTN network users and calls between SIP-based IP telephone network users.
Two comparison
In general, BICC is proposed directly to the application of telephone services, from traditional telecom camps, more rigorous architectures, so it can provide very good business in an existing circuit exchange telephone network in NGN Transparency. In contrast, the SIP architecture is not perfect as BICC definitions, and SIP is mainly used to support multimedia and other new services, and have more flexible and convenient features based on multiple business applications based on IP networks.
When using a BICC architecture, you can keep all the current functions, such as numbers and routing, etc., still use routing concepts. This means that the management method of the network is very similar to the existing circuit switched network.
And if the SIP-T architecture is used, the situation is different. From a routing perspective, there are two possible ways to introduce SIP-T in NGN: One is to keep routing concepts, that is, there is no meaning in the SIP environment; the other is to change the route to cater to the SIP environment.
In the first case, the call server, number, routing analysis, and signaling and the interoperability of the business are kept constant. Routing analysis guides the addressing of the target IP address, and the normal ISUP message package is transmitted in the SIP message. In this case, SIP-T can be regarded as a new protocol that adds package information on the ISUP. The second case is based on the Enum (IETF's Phone Number Mapping Working Group) database. In this way, the call control of the call server is completely different compared to call control in the existing circuit switched network. There will be no numbers and routing in call control, but business maps and interoperability are still required. Since the circuit identification code CIC, ISUP management process, message delivery protocol MTP, the standard ISUP protocol should be modified accordingly. The process of processing burst events, such as the mismatch, restart, overload processing, and the like of SIP and ISUP messages. The management of the network is simplified to some extent (if you don't need to build a signaling network, there is no routing definition). In addition, compared with existing networks, operators have changed huge changes in the control of the network and the control mode. Through the above analysis, we can see that the features of some existing telephone networks are lost to some extent by using the SIP-T protocol. To introduce these features, you need to expand the SIP-T protocol. In comparison, BICC can basically provide all existing telephone networks. I believe that the modified and standardized SIP-T can reach the support capabilities of BICC on traditional services, but it needs to be clear that which network principle is applied, which "routing" processing method is applied.
