1. The following is the test matrix 1 (the problem is not resolved ):
CalleeCaller
Jain-Sip-UA
X-Lite
IP-video
Jain-Sip-UA
It can communicate normally, but the number of frames is not enough and it is not smooth.
When the call is received, the xlite is dropped.
After the IP address is answered, "keep" is displayed, and UA audio and video can be received.
X-Lite
video servers are the leading integrated SIP network video products, requiring only a simple dial-up to remotely view the monitor area from the SIP video phone and enable two-way voice calls to the scene. And if alarm output is set to the SIP video phone, once the alarm event occurs in the monitoring area, you can watch the scene from the far end immediately. It
is called an SDP message.The problem is that the SIP terminal (UA) may not know anything about Nat. Therefore, IP addresses contained in SDP usually use internal IP addresses, that is, IP addresses known by the SIP terminal. In this way, when the communication peer wants to communicate with the SIP terminal, it will view the IP address in the SDP message, but no
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Core sip documents
RFC 2543
SIP: Session Initiation Protocol (obsolete)
RFC 3261
SIP: Session Initiation Protocol
SDP Related Documents
RFC 2327
Session Description Protocol (SDP)
RFC 3264
An offer/Answer Model with
Firewallnat
In fact, the essence of the light-mouth board is equivalent to NAT device. To achieve NAT penetration, there is no problem with the SIP processing of the optical orifice plate.
Firewall is a passive network security defense technology, located at the boundary of the network, executing the access control policy between the two networks, preventing the external network from illegally accessing the internal information resources, or preventi
Analysis of SIP route and record_route/SIP Routing Mechanism from: [url] Routing
Analysis of SIP Routing Mechanism (zz)We have already introduced sip-related knowledge about important SIP header domains, registration processes, and session processes. Now we will introduce t
This tool is a real-time analysis of the SIP Communication Protocol, which is the main function of the Communication Engineering of zhongzheng University. In addition, the distributed design can be used to analyze cross-domain sip packets, draw a complete signaling flowchart, and control the web interface... background in view of the fact that there is no free communication protocol analysis software for th
SIP (Session Initiation Protocol) Conversation Initiation Protocol is an application layer control (signaling) protocol for network telephony and conferencing. It is mainly a multimedia communication protocol based on IP network. It can realize the signaling functions are also using RTP as a media transmission protocol. Originally presented by the IETF Mmusic (multiparty multimedia session control) workgroup.
SIP
Address: http://www.sipcenter.com/sip.nsf/html/Chinese+SIP+Overview
Introduction
Next-generation services
Historical Review
Advantages of SIP: Web-like scalable open communication
SIP session Composition
Introduction
Communication providers and their partners and users are increasingly eager for a new generation of IP-based services. Now we have
Introduction
Communication providers and their partners and users are increasingly eager for a new generation of IP-based services. Now we have the SIP Session Initiation Protocol. SIP was born less than a decade ago in the computer science laboratory.
It is the first protocol suitable for multi-user sessions in various media content. Now it has become the specification of the Internet Engineering Task Gro
/success (actual Sipprovider callback), the Fail/sucess interface of the transactionclient is called Public classTransactionclientextendsTransaction {Transactionclientlistener transaction_listener; Publictransactionclient (Sipprovider sip_provider, Message req, Transactionclientlistener listener) {Super(Sip_provider); Request=NewMessage (req); Init (Listener, Request.gettransactionid ()); //This.transaction_listener = listener;} //The actual Sipprovider callback Public voidOnreceivedmessage (Si
Basic SIP Application
As one of the main VOIP communication protocols, the SIP protocol is simple, flexible, and open, and is gradually dominant in the VOIP communication field. The main methods used for SIP Communication include SIP terminals, proxy/targeted servers, location servers, and PSTN gateways. Currently, th
Author: gnuhpcSource: http://www.cnblogs.com/gnuhpc/
1. How is the SIP coming and how is it constructed?
In general, SIP is a lightweight signaling protocol, which can be used as a signaling for audio, video, and timely information.
Speaking of how the SIP is coming out, we need to mention H.323, and the standard has to mention ITU-T, we will first talk about t
1. Introduction to the SIP protocol
Many applications on the internet need to establish and manage a session. The session here refers to the exchange of data between participants. Considering the actual situation of participants, the implementation of these applications is often very complicated: Participants may move between agents, they may have multiple names, the communication between them may be based on different media (such as text, multimedia,
In a conversation, there are other correlations that will be sent. A conversation is an end-to-end sip relationship between two users that lasts a certain amount of time. The dialog process requires that the information between the two user agents be ordered and that the request be routed properly. In this specification, only invite requests can be used to establish a session. When a UAC makes a request in a conversation, it follows not only the gener
1 , what is SIPSIP (Session Initiation Protocol) belongs to the IP Application layer protocol, which is used to provide the user with session application on the IP network. A session refers to the communication of voice, video, and other media forms between two or more users, possibly IP telephony, conferencing, instant messaging, and so on.SIP is a signaling protocol that corresponds to a call signaling protocol (such as SS7 ISUP) in a traditional telephony network. Building a complete multimed
In the dialog, other related information will be sent. A conversation is an end-to-end sip relationship between two users that lasts for a certain period of time. The dialog process requires that the information between the two user proxies is ordered and the request is transmitted through the correct route. In this specification, only the invite request can be used to establish a session. When a UAC sends a request in a dialog, it not only follows th
RTP/RTCP/RTSP/SIP/SDP RelationshipRTP (real-time transport protocol, Transport layer)Real-time Transport Protocol) is a Transport layer protocol for multimedia traffic on the Internet. The RTP protocol details the standard packet format for transmitting audio and video over the Internet. RTP protocols are commonly used in streaming media systems (with the RTCP protocol), video conferencing and a Push-to-talk system (with either a/p or
SIP is a tool for generating C + + interface code for Python, similar to SWIG, but using a different interface format. The idea originated in Swig, primarily to create the QT package for Python, which was used to create PyQt and Pykde , and to support the QT Signal/slot system. This article mainly introduces the compilation and installation of SIP and the birth of C + + code into Python under the window pla
http://blog.csdn.net/noiile/article/details/115436
What are the problems with SIP from private network to public network?
Address translation of the package.
SIP address Translation inside the SIP message.
The RTP address translation in the SDP inside the SIP message.
The existing structure of the network is complex
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