This article from csdn ucser, http://blog.csdn.net/perfectpdl reprinted to indicate the source, thank you!
I have created a freeswitch learning and communication group, 45211986. welcome to join.
Freeswtich can be used as the rtmp and SIP gateway of the Streaming Media Protocol. It can communicate with the SIP video phone through flash in a web browser. This function can be used on the browser side f
Currently, the main reason for the combination of the codec and call control of the SIP protocol stack is the reuse of larger particles. In this case, the SIP is too bloated. It is not easy to expand.What I want to consider now is to break down the SIP library into two databases, one responsible for codec, and decoded by the
1. Authentication and encryptionThe role of certification (Authorization) is to show who you are, to prove to others who you are. The associated concept is MD5, which is used for authentication security. Note that MD5 is just a hash function and is not used for encryption. Because the hash function after processing the data can not reverse recovery, this way others can not steal your authentication identity password.Encryption (encryption) is the role of the data to be transferred to the process
SIP response code
The response code is included, and the HTTP/1.1 response code is extended. Not all HTTP/1.1 Response codes are properly applied, but only pointed out in the discount. Other HTTP/1.1 Response codes should not be used. In addition, SIP also defines a new response code series, 6xx.
1. Temporary response 1xxA temporary response, that is, a message response, indicates that the other server is
The integration of unified communication involves software, hardware, mobile devices, and fixed devices. The standards and protocols are different. Therefore, the integration is difficult.
In terms of IP phones and desktop applications, there are mainly SIP and H.323 Protocols; mobile videos are H.324M; Monitoring also includes H.323, SIP and proprietary protocols as well as MEPG2, MEPG4, and H. 264 and ot
In Windows Phone, when the input box gets the focus, the soft Input Panel (SIP) is automatically displayed for the user to input. When we click the physical rollback key, the SIP will be automatically hidden. So what event is triggered? Where should we write code for other operations?
We can register the keyup event in the input box. When the input box obtains the focus and click the back button, the syste
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,
From
The From header field contains the logical flags of the requesting initiator, which may be the user's Address-of-record. Just like the To header field, the From header field also contains a URI and can contain a displayed name (SIP display info).
To
The To Header field is the first and also the "logical" receiving place that specifies the request first ("First" is because it may refer to another receiving place),
Or the Address-of-record of the
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.
Csdn lidp http://blog.
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)
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 );
Kamailio is an open-source SIP server, formerly known as OpenSER. It runs C Programs on Linux/Unix platforms. It has good performance, flexibility and security. Weblinks · Homepagewithnewprojectname: http://www.kamailio.org · Home
Kamailio is an open-source SIP server, formerly known as OpenSER. It runs C Programs on Linux/Unix platforms. It has good performance, flexibility and security.
Web links
· Home
From: http://brekeke-sip.com/bbs/viewtopic.php? P = 11824 SID = 1337c4d609517c9d1f0fcc5167d7d5a1
1) Please go to Ondo SIP Server admintool> [config] menu> [system].Set [Java VM arguments] =-xrs
2) If you are also using Ondo PBX, please go to Ondo PBX admintool> [Options] menu.
Please find two [Java VM arguments] fields in the page.One in PBX system settings and one in media server system settings.
Please set[Java VM arguments] =-xrs
3) please go to
response retransmits
Timer E
Initially T1
Non-invite request retransmit interval, UDP only
Timer F
64 * T1
Non-invite transaction timeout Timer
Timer g
Initially T1
Invite response retransmit Interval
Timer H
64 * T1
Wait time for ACK receept
Timer I
T4 for UDP
Wait time for ACK retransmits
Timer J
64 * T1 for ud
Wait time for non-invite request retransmits
Timer K
T4 for UDP0 s for TCP/sctp
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