SIP protocol and IPLinkTM in next-generation network
Overview SIP review SIP and H.323 comparison SIP network module iPlink conclusions further information
■ Overview Based on the standard Internet protocol, SIP protocol (Session Initiation Protocol) is quickly popular in applications that wish to provide new services, communication, and network service providers (ASPS, CSPS, and NSPS). Another key product that provides the next-generation network powerful feature is IPLink, which is a standard software and hardware development platform to develop Internet protocol phone servers applications. IPlink not only provides interfaces with other Intel Dialogic boards, but also provides programming interfaces integrated with other telephone technologies. Whether it is a company or a telecom operator, computer and website developers can access telephone resources like accessing other Internet resources, so SIP greatly expands their ability to access telephone resources. The length of the SIP includes simplicity and relaxed sets of other standard IP protocols include HTTP, DNS, and SDP. SIP design decisive, which is more efficiently connected to and hanging channels than other protocols. Since SIP uses a standard IP architecture, familiar with the personnel programming is easy to accept and master it. Like IPLINK and SIP provide a simple interface. A PCI or CPCI's IPLink board can provide maximum flexibility in supporting standard IP call control, media gateway protocol, and speech coding algorithms, and complies with all relevant IP protocol specifications. SIP and IPLINK provide a solid foundation for developing next-generation network services, which are required for each successful ASP, CSP, and NSP. The combination of SIP and IPLINK is a key module for the "killer" application in the next generation network. If you want more to learn more about SIP, IPLINK, and open a next-generation network, please visit http://www.dialogic.com. Figure 1 Path of call control in next-generation network ■ SIP review SIP protocol is used to establish different types of dialogue between communication devices, media gateways, and media servers. When establishing a dialog, use a group of plain text messages to pass the IP address, port, media capabilities, encoding format, etc. At the end of 1999, it was proposed as a standard (RFC 2543) by the IETF Standardization Task Force. The establishment of the SIP protocol mainly draws on two web browsing and email protocols, HTTP protocols, and SMTP protocols. The first SIP produced in the IETF MMUSIC Workgroup, which primarily studies multimedia dialogue control. Refers to the distribution, management and coordination of multiple conversations, and is interactive with a variety of media (e.g., speech, image, and application applications) between multiple users. The MMUSIC organization is to design and refine three protocols that implement these features, and to ensure the compatibility of dialog-level compatibility in different conferences. These three protocols are:
SIP Session Description Protocol (SDP), today is mainly used by SIP and MGCP protocols to be used in Remote Function Call (RFC 2974) but there is no extensive applicable SIP protocol in RFC2543 continues to be updated, they Yes (basic) forward is compatible. Updates to Sip Are Continuing In The RFC 2543BIS, WHICH IS A (Nearly) Backward-Compatible Version of Sip. Moreover, IETF's SIP Working Group also defines a method of encapsulating ISDN User Part (ISUP) in SIP creek. . This method is also referred to as other techniques SIP (SIP for Telephony, referred to as SIP-T) This article is just about SIP's overview, and does not overwrite the details of all SIP. If you want to know more about SIP, please visit the website below: Henning Schulzrinne's Sip Site (http://www.cs.columbia.edu/ ~hgs/sip), SIP Forum Web Site (http: // www. Sipforum.org). Figure 2 A typical SIP message stream ■ Comparison SIP and H.323 have significant advantages and disadvantages. Advantages of H.323 Although H.323 is only designed to transmit voice and video on IP, it has been widely used. H.323 is a complete set of protocol stacks to transmit standard telephone voice services in IP and package switched networks. In addition to independent of other standards in design, the basic functions of H.323 include many phones, such as conference and call transfer. The advantage of this implementation is that it has left a clear interface to developing senior features and services, which guarantees high compatibility. The entire industry has made a lot of work in adding functions and promoting interoperability for H.323. H.323 will become an important signaling protocol in the next generation network. SIP's advantage is compared, SIP does not support advanced features such as conference and mute. The biggest advantage of SIP is that it is very simple. Unlike H.323 has a complete set of its own protocol stack, SIP mainly depends on protocols similar to RTSP and HTTP. SIP is more efficient than H.323 in terms of establishing and hanging calls, and there is less news. Moreover, SIP does not provide support for buttons. DTMF or is sent in the media stream (when using G.711 encoding) or by a special RTP package (when using G.726 or G.729 encoding). A unique function of SIP is an Invite message to be sent to multiple destinations at the same time. The RTP stream is then established with the distal end of the first reply OK message. Microsoft * NetMeeting client uses H.323 is a pivotable event. This allows users of most Windows * to use H.323. Similarly, Microsoft recently announced that SIP is supported in Windows XP. The SIP client will also get more and more. Table 1 Compare SIP and H.323.
SIPH.323 Message Coding Format Pure Text ASN.1 Call Establish Minimum Message Number 221 Call Establish Maximum Message Number 481 Processing DTMF Use RTP Protocol Process, can be included in the belt or with a special package (OUT-OF) -BAND) or in-band (recommended using an external RTP RTP) telephone transfer (AliaSing) proxy server or redirect server gateway or name server
1 Packets are larger than SIP, the number of bytes of the overall interaction is determined according to the ability of the H.323 terminal.
Figure 3 Modules in the SIP network ■ MEDIA GATEWAY PSTN network and IP networks require a call to support. The media gateway can be implemented in a variety of protocols. The example of Figure 3 is a PSTN-SIP media gateway. This king gate includes the interface of PSTN and IP, including the DSP for processing the call. DSP resources are mainly working. First, some PSTN protocols, such as the follow-up signaling (CAS) of E-1 and T-1, requires the detection / generation function of the key tone. Second, in terms of IP, it is also necessary to convert high bandwidth encoding G.711 to low bandwidth encoding, such as G.729A. Media Server Many voice services are provided by the media server. For example, a customer calls an e-commerce website that will be connected to an automatic voice response (IVR) or on the auto. These applications use speech, speech recognition, and audio, etc., are running on the media server. In the implementation of the next generation network, the media server is like another phone (or said, a terminal). Therefore, they can enjoy the advantages of the same SIP as the user. The media server provides all voice computing resources required to interact with the call party. Now the network is completely-oriented, embedded DSP resources allow programming of voice flow, and developers can provide the required voice, audio, and speech recognition of speech, audio, and speech recognition for their specific services. Media servers allow technology developers to provide or provide or with the latest resource connections, such as textual conversion engines (TTS), speech recognition, echo, noise reduction, etc. A main feature of the agent, redirection, and location server SIP is that it separates a logical address of a user and his actual (physical) address. This allows the user to define a constant logical address and then use it to map or alias to one or more of the actual addresses. This feature is provided by proxy, redirect and location servers. In Figure 4, in the SIP network, the proxy server and registration / location server combines users to identify themselves with an address, and actually calls to one or more different locations. Typical processes for SIP calls using a proxy server, for example, Joe Smith's address is jsmith@sip.org. People who want to contact Joe can initiate a SIP call for that address. The proxy server will send the call to where the Call is determined according to the JSMith user. The SIP's INVITE message is sent to the "JSMith" set address. When the called party responds to the proxy server, the proxy server also forwards this response to the calling party. A RTP dialog is then created directly between the calling party and the called party. The proxy server will continue to participate in the processing of call control messages as needed, or exit message processing. In some cases, the system can make the system more done without using the proxy server. Figure 4 Proxy Server (Click to see the larger image) Use the redirected server in the typical flow of the SIP call for the server, only the first INVITE message sent to the called party and feeds back a special The response gives the calling party. The calling party removes a new address from the response and sends the INVITE message to this address. This address may be a real called party, a proxy server or another redirect server. Since then, all messages are passed directly between calling party and new address. Regardless of the proxy server or the redirect server, you need to understand the customer's true location to forward the call. This requires a location server that can be used with a small database on a machine, or using remote protocol, such as LDAP or WHOIS.
Typically, the SIP terminal will use the SIP Register message to register its contact information to the location server. For example, a user uses his email in the location server as joe.smith@sip.org, when he is at home, he can register the IP address of his home SIP phone, and when he is in the company When he can register the IP address of his company's SIP phone to the location server. Figure 5 Redirects the server (click on the big picture) The typical process of the SIP call using the location server When someone wants to contact Joe Smith, use Joe.Smith to initiate a call to the Sip.org agent or redirect the server. The proxy server will contact the location server and the location server finds from the data air to which address sends a call. The proxy server then sends the INVITE message to the address of the called party and waits for a reply. If the server plays a redirect server, it uses the location server to find the called party address and send this address to the calling party in the redirected message. The calling party will send the message directly to the called party. In both cases, the RTP media stream is directly established between the calling called party. Many proxy servers also provide additional call processing when receiving or transmitting a call. When a call is sent to the proxy server, the user can set this call to different addresses based on different times in one day, or different responses. For example, the user can register three addresses, and the proxy server can call each address until the phone is answered. Or the server can call three addresses simultaneously, turn it on one or more calls and returns. These services can be used to do uniform number? Quot; one-number follow-me "program. ■ IPLink IPLink is an open, standard IP phone platform for gateways or other value-added services. A board card There are Ethernet interfaces and PSTN interfaces. Oem, application developers and integrators create next-generation network IP gateways and IP media servers for businesses and public.com. It has a high degree of flexibility, robust, is With many tests. Advanced encoding supports IPLink to support all standard IP codes, including G.723.1, G.729A, G.711, and enhanced encoding GSM-EFR. It uses separation call control and media The function of processing supports various IP call controls or media gateway control protocols, such as H.323, SIP, MGCP, and H.248.IPLINK are also interoperable with other VoIP programs, including Cisco, CLARENT, VOCALTEC, and DIGI. Protocol Architecture IPLINK protocol design allows developers to choose host-based signaling protocols or embedded signaling protocols. The host-based signaling protocol mode is also called "split call control", refers to the handling of call control and media processing The card is to do, its advantages are unparalleled flexibility. In this mode, the IPLink board controls the RTP stream, the host's application controls the call control protocol. There are two advantages: First, all The iPlink board in the system share a call control (different IP addresses for media flows). Second, developers can choose standard SIP or Megaco protocol, or choose non-standard or private protocols. Support and development tools except Open modules are used by developers, Intel also provides training, coordinating different manufacturers integration, providing a large number of exemplary programs with source code, as well as national compatibility tests. You can at http: //www.dialogic. The COM found a series of supports provided by developers, as well as users in the value chain, but also users of communication services. Intel provides a reference system for the next-generation network voice communication service.
Root root in traditional circuit switched networks, voice services need to spend the most effort to the next generation network. Traditional development processes begin to select a correct product. Developers need a test different solution. When the right product is found, they will start integrating it into the entire program. Generally, this integration has not added value, just a process that spends time experience. Intel provides a reference system that can avoid several months of testing, possible errors and integration into the scheme. These can help developers focus on providing their special value-added services. The reference system also contains Intel multi-year experience in finding the best development plan. The entire industry has developed in many small developers and the wonderful ideas of the development team. Intel has cultivated an innovative environment by providing answers that developers usually encounter. ■ Conclusion SIP has become more and more popular in ASPs and CSPs that wish to provide innovative new services. The open architecture of IP has high flexibility to create new services. Continuously introduce new user terminal devices, such as protocols like SIP, allowing developers' innovative capabilities unrestricted. They let the Internet reach or exceed all users expectations, which may be understood by understanding technology or just want simple communication. IPlink is a comprehensive standard software and hardware development platform that faces the IP phone server in the next generation. SIP and IPLINK can provide innovative services in the next generation of the next-generation network, which is pivotable for today's ASP, CSP, and NSP. ■ Further information I hope to learn about the voice communication service architecture of the Intel reference design. "Reference Systems for Next Generation Network Voice Services." This article analyzes the commercial goals and next-generation speech architecture of the next generation network provider. Based on a standard computing platform. You can download this white paper here http://www.dialogic.com/company/whitepap/7299web.htm. I hope to learn more about SIP and IPLINK and open-ended next-generation networks, please visit http://www.dialogic .com This document is related to the Intel product, this document does not include any intellectual property of the product. Unless it is provided in the name of Intel and for selling this product service, Intel is not responsible, and does not provide guarantees for issues encountered in the sales and / or use of products, including the applicability of a particular purpose, product Sales, as well as infringement of patents or other copyrights and intellectual property rights. Intel products do not apply to any application that may generate human injuries or deaths, such as medical, first aid or proven. Intel may always modify this manual without notice.
