SIP is an application control signaling protocol proposed by IETF. As the name implies, it is used to initiate a session. It can be used to create, modify, and end multimedia session processes attended by multiple participants. Participants can communicate with each other through multicast, unicast, or network connection.There are clients and servers in the SIP. A client is an application that establishes a
I believe that you have mastered the basics of IMS and SIP protocols and some precautions. These contents are also described in two articles, "briefly describing the SIP protocol in IMS" and "Problems Existing in IMS and SIP. Next we will explain the SIP Protocol extensions in IMS.
1.
parameters;
Call Establishment: the establishment of the session parameters of the caller and the called party;
Call Management: includes transferring and terminating sessions, modifying call parameters, and calling services.
The SIP protocol can be combined with other IETF protocols to establish a sound multimedia structure, such as providing real-time data transmission and service quality QOS) Feedback of real-time transmission protocol RTP) real-t
need to modify the Conditional compilation option. For Win32-incompatible APIs, you can replace and modify them with other WinCE APIs. In addition, some header files and pre-defined files of WinCE and Win32 are also different and need to be modified. table 1 provides some migration examples.
Porting JRTPLIB on WinCE is similar to porting the SIP protocol stack. Note that the maximum RTP loading data packets under WinCE is 2 K, which is different from
Previously, we handled some application problems related to the SIP Session Initiation Protocol. Many solutions are summarized. I wonder if you have mastered it. See Basic Q A of the SIP Session Initiation Protocol. Let's add more content in this article.
Is X standard supported?
We have implemented and tested a large number of standards. We listed these standards in the Product Information Center (see ref
Today the work encountered two devices between the SIP packet crawl and analysis, and then combined with the RFC3261 document description, recorded today understand.1.SIP Protocol:Detailed RFC documentation for SIP is visible: rfc3261Session initiation, which allows you to use Internet endpoints (user agents) to find participants and allow you to create a shareab
Today, I encountered the capture and analysis of the SIP packets between the two devices. Then, combined with the instructions in rfc3261 documents, I recorded what I understood today.
1. SIP protocol:The detailed RFC documentation for SIP can be found at: rfc3261Session Initiation (Session Initiation Protocol) allows the use of internet endpoints (User proxies
effective way to debug network programsThe following describes the software mainly based on open source software .... For details, see the original document.2. Video Communication Based on the SIP protocolHttp://tech.163.com/05/0101/15/1915T5RL00091590.htmlI talked about three aspects:The relationship between the SIP protocol and its development, the basic components of the
Chapter 3 SIP messages
SIP is a text-based protocol that uses the UTF-8 character set (rfc2279 ). A sip message can be a request sent from a client to a server, or a response sent from a server to a client.
Both requests (rfc3261 section 7.1) and responses (rfc3261 section 7.2) are in the format described in rfc2822. However, there are some differences between ch
VoIP intercom systems, the qutecom software interface is as follows:
Ii. Official qutecom website
Qutecom open source Official Website: http://www.qutecom.org/
As follows:
Iii. Introduction to sip and RTP protocols
3.1 sip Overview
SIP is an application control (signaling) protocol proposed by IETF [1]. As the name implies, it is used to in
The SIP Session Initiation Protocol has been added in many systems. Therefore, we have summarized and summarized some problems of the SIP Session Initiation Protocol, hoping to help you solve some problems. For the sip faq, the Session Initiation Protocol SIP Servlet 1.0 support (JSR 116) was added for the first time i
The registration process of SIP is easy to understand. First, I have a number, but if my number can be moved, how can the server find me? The SIP registration mechanism reports the location of the SIP terminal to the registration server. The registration server is only a logical role. It is not necessarily an independent physical entity. It can be the same physic
Kamailio is an open-source SIP server, formerly known as openser.
Kamailio is an open source, gpl2, SIP Server Routing platform. It is written in C for Linux/Unix plaforms and focuses on performance, flexibility and security.
On Nov 04,200 8, kamailio and SIP Express Router have started the SIP router project.
Web Link
1.
Sip1.1. overview 1.1.1.
Basic components of the SIP System
(1) User AgentIn sip, the user proxy (UA) is the endpoint entity. The User Agent initiates and terminates a session by exchanging requests and responses. As an application, UA includes the user proxy client and user prox
SIP: Session Initiation ProtocolDocument StatusThis document has developed an Internet standard tracking protocol for Internet communications that asks for discussion and elevation of recommendations. Please refer to the "Official Internet Protocol Standard" for the standardized status of this agreement. Forward Unlimited!Copyright noticeCopyright (c) Internet community (2002). All rights Reserved!ProfileThis document describes the Session Initiation
The business support environment mainly includes the application server, Business Management Server, and business generation environment. They work together to quickly complete the tasks that provide users with diverse and flexible value-added services based on the next generation network. The application server is the subject of the business support environment. The Business Management Server and business generation environment can be used as part of the application server.
Figure 2
"user parameter" field is added after the domain name. This field has two optional values: IP address and phone number. "Host" can be a domain name or IP address. "Port" indicates the port number to which the request message is sent. The default value is 5060. The password can be written into the sip url, but this is not recommended for security reasons. Other parameters are easy to understand or rarely used. We will not describe them here.
2) MGCP e
Project name
Description
Files
Activity %
SIPP
SIPP is a performance testing tool for the SIP protocol. its main features are basic sipstone scenarios, TCP/UDP Transport, customizable (XML based) scenarios, dynamic adjustement of call-rate and a comprehensive set of real-time statistics.
84.69%
Shtoom-a python sip framework phone
A software
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
protocol, it only uses the SIP defined by IETF in some way. therefore, there are also specific requirements for SIP in public mobile networks, such as low bandwidth, roaming, security requirements, service quality (QoS), and billing control.
In the SIP model, to establish a session, the user proxy client initiates a r
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