Use FireBreath to develop real-time playback interfaces (Yate + SIP + FFMPEG + SDL) and firebreathyate
At that time, such a blog post was really needed to guide this function module. Unfortunately, FireBreath has very little information on the Internet and is not very familiar with C ++, so we tried and explored it all the way. Fortunately, we have implemented this module, and now we have recorded it.
First of all, our Yate
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 );
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
Mysql-Proxy proxy configuration, mysql-Proxy proxyI. Principles of Mysql-Proxy
Mysql-Proxy is a simple program between your client and Mysql Server. It can monitor, analyze, and change their communication. It is flexible and unrestricted. Its common uses include load balanci
I began to study the VOIP/SIP agreement from 09, open source project also saw a few, the earliest Pjsip 05 began to push the time, began to pay attention to, also in their own winmobile project used. Later also saw Sipdroid,imsdroid (Doubango), Linphone,csipsimple (PJSIP).I think the best advantage of Linphone and Csipsimple,linphone is the full platform support, Android,ios,winphone,windows,linux,mac osx,web all support, but the quality is still unde
IOT command (based on sip) client API design for java, iotsipThe Iot Device Control we implement is implemented by extending the sip protocol. Because pjsip is implemented based on pjsip, while pjsip uses C Programming, how to make the business layer (android end, java) easier to use the provided command API is the focus, the original method is to encapsulate (C ---> jni ---> java) from the underlying c, wh
streams have a public media format 415 Response ( Media type not supported ) , and join 304 Warning Header field ( Media type not supported ) . 3 . Multicast Operations( 1 ) The multicast addresses that are accepted and sent are the same. ( 2 ) is called not allowed to change the media stream only hair, just accept or receive / To the hair characteristics. ( 3 ) If the call does not support multicasting, the loopback - Response and the Warning ( multicast not available ) . 4 . delayed Media
I recently found out doing a sip capture on a FreeBSD system is a little different than centos or other Linux distributions. here is a easy to use command that will grab the SIP packets from TCP dump. this will give you an easy to read text file for debugging or tracing.
> Tcpdump-I bce1-n-S0-vvv UDP port 5060>/usr/src/capture_file
Let's go over the options for this command:
-I = interface which on my BS
P2p-sip is a peer-to-peer telephone protocol, and someone wrote a python implementation.
This only supports python2,2.6 above
PIP installation, or download installation package decompression.
After decompression has the readme, chews the English.
Write webcaller.py
Import gevent, sys from gevent import monkey; Monkey.patch_all () from GEVENT.PYWSGI import wsgiserver to CGI import Parse_qs, escape import logging from logging Impor
T config logging.co
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
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
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