In my previous article, I introduced how to use a SIP Soft Phone for direct calls. However, if many other users need to talk to each other, at the same time, authentication and control of User Account logon is required based on security considerations, in these cases, you need to establish a sipserver.Role of sipserver:Call control and processing functions, business provision/support functions, user management functions, protocol processing functions,
... 13
8.6.2.1 when the communication parties are at different levels of Nat... 14
8.6.2.2 related to the NAT type... 15
8.6.2.3 other cases... 16
8.6.2.4 peer reflexive in Internet p2p... 16
9 Application of ice in SIP... 16
9.1 both parties collect three groups of addresses ...... 17
9.1 A sends invite to B... 18
9.2 B returns 100, 101, 180 to a... 18
9.3 B returns 200 OK to a... 19
9.4 A returns ack to B... 19
Last week I wrote 1st, 2, 3, 4, and 5
I am responsible for custom development of the SIP/IMS video client and support access to the SIP Soft Interface.Switch, IMS core network, supportedVoice, video, and instant messaging functions. The video formats support h263, h264, and MPEG4 soft encoding solutions. The hardware coding/decoding interface is provided for interconnection and servers. If you are interested, contact me.
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Haha. I 've talked about some SIP application scenarios. I will write some theoretical and boring things in the future. Before that, let's continue to make it easy to tell you a story about how to learn HTTP.
At the very beginning, I took an HTTP book first. In short, it was a brick book. After reading the two chapters, we found that we should use the knowledge of TCP/IP. So let's look at TCP network programming. Two months later, I still don't kno
After OS X upgrades to El Capitan, it provides a security-related pattern called SIP (System Integrity Protection), also known as rootless mode, which is a new feature that emphasizes security for OS X, which prohibits the software from being used as root in Mac running on, upgrade to OS X 10.11 Maybe you'll see some apps are disabled so that/usr/bin folders we can't read and write properly, but it also causes some programs (such as homebrew and Git)
SIP Reply Message Status codeand featureType Status Code status descriptionTemporary response (1XX) Trying is in processRinging ringing181 call being forwarder is forward182 Queue Queue181* Session Progress SessionsSession success (2XX) OK session succeededRedirect (3XX) multiple multiple options301 Moved Permanently permanent mobile302 moved temporaily temporary movement305 Use Proxy User agent380 Alternative service Replacement services Request fail
650) this.width=650; "title=" clip_image002 "style=" border-top:0px;border-right:0px;background-image:none; border-bottom:0px;padding-top:0px;padding-left:0px;margin:0px;border-left:0px;padding-right:0px; "border=" 0 "alt = "clip_image002" src= "Http://s3.51cto.com/wyfs02/M00/84/57/wKiom1eNzEnCj0QxAABdCgY5914328.gif" height= "384"/>The difference between H323 and sipSIP P2p:trunkSIP C/S: End pointSIP dialing behavior does not support KPML. Every keystroke is sent once.The default
In many cases, the SIP does not go directly to the target host, but goes through many intermediate node servers. In the request message, the Via header field indicates the nodes that have passed through (each node passes through, add a via header). In the response message, the via header field indicates the node that the message will go through next (each time the request is returned from the original path, a via header is deleted from each node ).
T
Prack English translation (the provisional Response acknowledgement), you can call IT security information! This compares the image.The final response in the SIP is understood to be reliably transmitted, such as a 200OK response to the invite, and UAC will give an ACK telling UAS that it has received 200OK. The reliability between 200 and ACK is end-to-end. Prack is a mechanism for guaranteeing the reliable transmission of temporary messages (101-199)
In PPC development, you sometimes need to hide the sip. There are many ways to hide the SIP in Windows Mobile 5.0. The following are several methods:
1. shsippreference (m_hwnd, sip_down );
2. sipinfo Si;
Memset ( Si, sizeof (SI ));
Shsipinfo (spi_getsipinfo, 0, Si, 0 );
Si. fdwflags = ~ Sipf_on;
Shsipinfo (spi_setsipinfo, 0, Si, 0 );
3. shfullscreen (hdlg, shfs_showtaskbar, shfs_hidesipbutton );
The concept of unified communication needs to be understood from the combination of multiple protocols. Among them, there is more participation in the SIP protocol. The integration of unified communication involves software, hardware, mobile devices, and fixed devices. The standards and protocols are different. Therefore, the integration is difficult.
For IP phones and desktop applications, the main protocol is S
Some friends may be familiar with the SIP protocol. In this regard, the most prominent thing is the VoIP service. In VoIP services, the SIP protocol and the SIP server are often involved. Next, let's take a look at the traversal problem on the SIP server.
1. Description of SIP
Following last night. decided to update the SVN that comes with the system. The SVN version number that comes with it is 1.7. Crossing Network Svn:http://www.wandisco.com/subversion/download#osx The latest version number is 1.9.13, decided to upgrade under.Unexpectedly due to EI Capitan sip problem Toss a good freshman meeting. Would not have wanted to record. But because sip this egg-ache thing decides sti
If you have a SIP account from a carrier, you can configure the SIP to dial an external phone. The SIP account (or the device providing the account) is referred to as the SIP gateway in FreeSWITCH. Adding a gateway only needs to create an XML file in conf/sip_profiles/external/, with a name that can be randomly used, s
Answer code
The answer code is included, and the http/1.1 answer code is extended. Not all http/1.1 answer codes are properly applied, and only the appropriate ones are indicated in the fold. Other http/1.1 answer codes should not be used. Also, SIP defines a new answer code series, 6xx.
1 Temporary answer 1xx
A temporary response, a message-nature response, flags that the other server is processing the request and has not decided on the final answe
1, added SIP Provider, add the configuration file in Conf/sip_profiles/external
This SIP Provider requires REGISTER, and since FreeSWITCH is accessed through NAT, it is set to send pings for 30 seconds.
2, the did mapped by this SIP Provider to the corresponding extension
SIP Profile External.xml Sets the context d
At BEA, my role is to help build and support ISVs of applications on WebLogic Communications Platform to build an ecosystem.
It is easy to use WebLogic SIP Server to compile an aggregate J2EE/HTTP/SIP application. This article details the architecture behind a complete conference application that uses Cantata's Media Server to stream audio and video.
The two smart developers took less than a month to comp
SIP, which has always been known as "simple", is not so simple, but it is difficult to grasp anything.This document is designed to keep track of the various doubts and problems encountered during SIP usage.First, Response 422 Session Interval Too SmallThe invite messages sent are as follows:INVITE SIP:806@192.168.8.11sip/2.0Via:sip/2.0/ws 9srpbdc87v1s.invalid;bra
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