Comparison of H.323 and SIP
At present, 3GPP regards SIP as the core protocol of the third-generation mobile communication all-IP network. The protocol formed by NetMeeTIng in the Windows XP operating system is also changed from H.323 to SIP. Taking into account the flexibility of its business, SIP protocol will become The direction of future development.
Comparison of SIP and BICC
BICC is directly proposed for the application of telephone services. It comes from the traditional telecommunications camp and has a more rigorous system architecture. Therefore, it can provide good transparency for the implementation of services in the existing circuit-switched telephone network in softswitch. In contrast, SIP's architecture is not as complete as defined by BICC. SIP is mainly used to support multimedia and other new services, and has more flexible and convenient features in multi-service applications based on IP networks.
BICC is developed on the basis of ISUP. It is relatively mature in terms of voice service support and can support all narrowband voice services, supplementary services and data services in the past, but the BICC protocol is complex and has poor scalability. In wireless 3G applications, the BICC protocol is at the Nc interface of the 3GPPR4 circuit domain core network, providing support for call connection between (G) MSCServers.
In fixed network softswitch applications, the BICC protocol is in the call control layer of a layered architecture, providing support for call connection between different softswitches. When the BICC system architecture is adopted, all current functions can be kept unchanged, such as number and routing analysis, and the routing concept is still used. This means that the management of the network is very similar to existing circuit-switched networks.
SIP is relatively less mature than BICC in terms of voice services, but it can support strong multimedia services and has good scalability. According to different applications, it can be expanded accordingly. In fixed network softswitch applications, the SIP protocol is in the call control layer of a flat architecture, providing support for call connection between different softswitches. When using the SIP architecture, from a routing perspective, there are two situations:
In the first case, the normal ISUP message adds some information and is encapsulated in a SIP message for transmission. The functions of the call server, number, route analysis and signaling, and service interconnection remain unchanged. Route analysis guides addressing to the target IP address .
The second case is based on the ENUM (IETF Telephone Number Mapping Working Group) database. In this way, the call control of the call server is completely different from the call control in the existing circuit-switched network. There will be no number and route analysis in the call control, but service mapping and interworking are still required. Since the circuit identification code CIC, ISUP management process, and message transfer protocol MTP are not used, the standard ISUP protocol needs to be modified accordingly. The management of the network has been simplified to some extent (if there is no need to build a signaling network, there is no route definition). In addition, compared with the existing network, the operator's control of the network is reduced, and the control method has undergone tremendous changes.
Through the above analysis, the use of the SIP protocol will lose some functions in the existing telephone network to some extent. To introduce these functions, you need to expand the SIP protocol. In comparison, BICC can basically provide all the functions of the existing telephone network. I believe that the modified and standardized SIP can achieve BICC's ability to support traditional business.
Comparison of SIP-T and SIP-I
Regarding the interworking of the softswitch SIP domain and the traditional PSTN, there are currently two standard systems, namely the SIP-T protocol family of IETF and the SIP-I protocol family of ITU-T.
1. SIP-T protocol of IETF
SIP-T (SIP for Telephones) is defined by RFC3372 of the IETFMMUSIC working group. The entire protocol family includes RFC3372, RFC2976, RFC3204, RFC3398 and so on. It adopts an end-to-end research method to establish three interworking models when SIP and ISUP intercommunicate, namely: the call is initiated by the PSTN user via the SIP network and the PSTN user terminates; the call is initiated by the SIP user and the PSTN user terminates; Terminated by SIP users.
SIP-T proposes two methods for the interworking of SIP and ISUP, namely encapsulation and mapping, which are defined by RFC3204 and RFC3398, respectively. However, SIP-T only focuses on the interworking of basic calls, and basically does not involve supplementary services.
2. ITU-T SIP-I protocol
The SIP-I (SIPwithEncapsulatedISUP) protocol family includes TRQ.2815 and Q.1912.5 of the ITU-TSG11 working group. The former defines the technical requirements for the interworking between SIP and BICC / ISUP, including the interworking interface model, the protocol capability set that the interworking unit IWU should support, and the security model of the interworking interface. The latter defines the interworking of 3GPP SIP and BICC / ISUP according to the different protocol capability configuration sets that IWU needs to support on the NNI on the SIP side. In general, the interworking of SIP and BICC / ISUP, when SIP has ISUP message encapsulation SIP-I) Interoperability with BICC / ISUP, etc.
The SIP-I protocol suite reuses many IETF standards and drafts, covering not only basic call interworking, but also BICC / ISUP supplementary service interworking.
3. Comparison of SIP-T and SIP-I
Obviously, the content of the SIP-I protocol suite is far richer than that of SIP-T. The SIP-I protocol family includes not only basic call interworking, but also CLIP, CLIR and other supplementary services; in addition to call signaling interworking, resource reservation and media information conversion are also considered; existing fixed networks In the softswitch environment, the interworking of SIP and BICC / ISUP, and the interworking of mobile 3GPP SIP and BICC / ISUP, and so on.
What is particularly important is that the SIP-I protocol suite has the clarity, accuracy, detail, and specificity inherent in the ITU-T standard, and is highly operable. And 3GPP has adopted Q.1912.5 as the final standard for interworking between IMS and PSTN / PLMN. Therefore, the interworking between the softswitch SIP domain and the PSTN should follow the ITU-T SIP-I protocol suite. In fact, many telecom operators have finally chosen SIP-I and abandoned SIP-T. Based on the above comparison and analysis, we can draw the following conclusions:
In terms of interoperability protocols between softswitches, SIP-T is the most commonly used in fixed networks, and BICC is the mobile application. The future development direction is SIP-I; in terms of control protocols between softswitches and media gateways, MGCP is more mature, but H.248 inherits all the advantages of MGCP, and there is a trend to replace MGCP; in terms of the control protocol between softswitch and IAD, MGCP is more mature, but H.248 inherits all the advantages of MGCP, there are The trend of replacing MGCP; SIP is the trend in the control protocol between the softswitch and the terminal; SIP is the mainstream between the softswitch and the application server, and currently the service interface is basically mature; the application server and the third-party business, Parlay is the direction, But commercial use is not mature at present.
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