sip isup

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

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

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

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-

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC clients establish a call through interactive candidate to achieve NAT traversal, if these interactions candidate in the Offer/anwser SDP will lead to through the establishment time is very slow,Because the WEBRTC client needs to communicate with the Stun/turn server to get these candidate, the entire process is block, in order to increase the speed of call establishment, WEBRTC introduced trickle ice, in in

Preliminary Exploration of OsipToday, we started to study the SIP protocol, called Session Initiation Protocol, which is a required protocol in VoIP.First, I found the RFC document, rfc3261, and more than 200 pages of English documents, which is too slow to read. Later, I found it was said that it was Huawei's internal sip training materials, or the Chinese language was good, and I had a general understandi

Find this Linphone when looking for Open Source SIP Phone reference, download: http://www.linphone.org/eng/download/packages/linphone.html After reading about the structure, we used Osip, exosip, and ortp protocol stack for development. 264 of the support was x264 (a sub-project of VLC ). Haha, I had a lot to do with what I used to do. Basically, I used to develop the SIP protocol stack based on Osip, exo

* RFC 2327 (SDP: Session Description Protocol) * RFC 2782 (dns srv Resource Record) * RFC 3261 (SIP: Session Initiation Protocol) * RFC 3263 (locating sip servers) * RFC 3264 (an offer/Answer Model with SDP) * RFC 3265 (SIP-specific event notification) * RFC 3266 (support for IPv6 in SDP) * RFC 3403 (DNS naptr Resource Record) * RFC 3420 (Internet media t