sip communicator

I have developed custom Sip/IMS video clients, and supported voice, video, and instant communication functions. The video formats support h263, h264, and MPEG4 soft decoding, and provide hardware coding/decoding interfaces, provide servers. If you are interested, please contact me. Registration process (Java --> C ++ --> C) Register (ngnsipservice. Java) | Register (ngnregistrationsession. Java) | Register _ (sipsession. cxx) | Tsip_action_register

The HTML5 SIP client is an open-source client that fully utilizes JavaScript to integrate social networking (Facebook, Twitter, Google +), online games, and e-commerce applications. No extensions, no plug-ins, or necessary gateways. The video stack technology relies on WebRTC. Like Demo Video demo on the home page, you can easily implement real-time video/audio calls between Chrome and IOS/Android mobile devices. This client is a technology that can

the Application Scenario (Call Center) of the author, the main problem to be solved is that the agent can traverse in the NAT environment and send information to the server. Because the SIP Soft Phone used by the agent is developed by our company, it can be ensured that rport and received are supported. 3.2 instance The following is an instance that sends the register information. The request's via header contains the rport parameter with no value,

In the process of this test, the most disturbing problem is that the gateway is not receiving the alerting event during the call, causing the state machine to be disturbed. In fact, the SIP protocol already defines the reliability of the temporary response. It is stipulated in the SIP standard that the definition of the reliable transmission of temporary messages during the call establishment process can be

The Cisco SIP VoIP architecture solution provides users with many services. The table lists various IP telephone services that can be implemented by Cisco SIP VoIP Architecture solutions.Cisco SIP VoIP architecture solution ComponentsThe Cisco SIP VoIP architecture solution consists of the following elements:Sip ip tel

The SIP protocol and some of its applications are worth learning. In this regard, we will explain the configuration of rtp sip today. SIP (Session Initiation Protocol) is usually used for VOIP calls, call establishment, call negotiation, and call termination. it helps two terminals to recognize each other, but it does not process media. When A call is established

former scenario usually involves early media, such as Ring-Back Tone (the music you hear when calling a person subscribed to this service) or interworking with PSTN. the later scenario may involve resource reservation. these scenarios, by definition, require setting and changing the media properties as the call begins, and this forces sip to take a complex path to support them. The following flow demonstrates a complex invite scenario. For clarity pu

Processing of FreeSWITCH SIP signaling in Mod_sofiaFirst, a thread that handles SIP messages is created inside the module's load (mod_sofia_load): /* Start one message thread /switch_log_printf (Switch_channel_log, Switch_log_info, "starting initial message Thread.\n "); Sofia_msg_thread_start (0); The Config_sofia function is then called. if (Config_sofia (Sofia_config_load, NULL)!= switch

Label:Recently studied asterisk configuration, before the SIP account is configured in the sip.conf file, manual writing dead, the current demand, is the dynamic new SIP account, saved in the database.After adding data to the database, use the command SIP show users to not load the SIP account to the database.1. Downlo

introduction: The previous time experienced the Xcode compiler code was injected into the event, this time mac os X elcapitan system upgrade, enabling a higher security protection mechanism: System Integrity protection systems Integritypro tection (SIP), is by Design? Or is it a coincidence? for System Integrity Protection systems Integrity Protection (SIP), you canAppleDownload the website to study, fro

Practice. It is not enough to know some knowledge. You need to practice it first. Now we have learned about the SIP protocol. Here we will share the practice process of a netizen's sip invite. I hope it will be useful to everyone. Request sent by linphone in sip invite (reguest) INVITEsip:to@192.168.105.14SIP/2.0 Via:SIP/2.0/UDP192.168.105.5:5060;rport;br

Research on ice-based sip signaling penetration over symmetric NAT technology Zeng Li, Wu Ping, Gao Wanlin, Wu wenjuan (Department of Computer Science and Technology, Agricultural University of China, Beijing 100083, China) 2 (School of information, Renmin University of China, Beijing 100872, China) Abstract what is one of the practical difficulties faced by IP-based speech, Data, video, and other services in the NGN network?Effectively Penetrate vari

Chapter 1 SIPP IntroductionSIPP is a tool software used to test the performance of the SIP protocol. This is a GPL open source software. It contains some basic sipstone user proxy workflows (UAC and UAS) and can be used to create and release multiple calls using invite and B ye. It can also read the XML scenario file, that is, the configuration file that describes any performance tests. It dynamically displays test running statistics (call rate, back-

HTTP Authentication SIP provides a stateless, trial-and-error mechanism for the authentication system. This mechanism is based on HTTP authentication. At any time, the proxy server or UA receives a request (except in section 22.1), which attempts to check the identity confirmation provided by the request initiator. When the sender confirms the identity, the request recipient should confirm whether the user has been authenticated. In this document, it

client should try again in the sip uri mode. If you receive a 420 (incorrect extension) Response (rfc3261 section 21.4.15 ), the request lists an extension option not supported by the proxy or uas in the require or proxy-require header domain. UAC should ignore the extensions listed in the response's unsupported header and try again. In all the above cases, the request is modified accordingly and a new request is created. This new request creates a n

Simple Solution: c: \ Python26 \ python.exe setup. py py2exe -- extends des sip It is very convenient to use pyqt to complete the form interface, but there will be problems after packaging it into exe. The solution on the internet is as follows:Another Solution to the same problem: from distutils.core import setupimport py2exesetup(windows=[{"script":"main.py"}], options={"py2exe":{"includes":["sip"]}})

According to the RFC3261-13.2.1, the offer/answer model used by the SIP is established in the dialog environment. The RFC also specifically imposes restrictions on offer/answer Interaction: 1.The initial offer must be in the invite message or the first reliable non-Failed response. Note: At that time, the reliability Effect of rfc3261 was only 2 **. Next we will talk about 1 ** (except 100. 2.If the initial offer is in the invite message, the answer m

Status Codes and type status codes of SIP response messagesTemporary response (1xx) 100 trying in process180 ringing181 call being forwarder call forward182 queue181 * session progress session Successful SESSION (2XX) 200 OK session successfulRedirection (3xx) 300 multiple multi-Choice301 moved permanently permanent movement302 moved temporaily temporary movement305 use proxy user proxy380 alternative serviceRequest failed (4xx) 400 bad request error