sip sdp

SIP protocol Learning 1The SIP protocol is an application layer control protocol proposed by IETF for multimedia communication over an IP network. A hierarchical method is used to create a service. A control protocol on the application layer is used to create, modify, and terminate multimedia session processes with multiple participants. Members participating in a session can communicate through multicast,

* 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 (supp

description, such as the requested media, bandwidth, or address type, are not received.The 606 (notacceptable) response means that the user wants to communicate, but cannot fully support the Session Description. 606 (not acceptable) a response can contain a reason list in the warning header to explain why the session description is not supported. The warning cause code is listed in section 20.43.In the response, a message body containing the media compatibility description can appear. The messa

Developed for SIP/IMS video clients, supports access to sip Softswitch, IMS core network, andVoice, 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. Implements standard-based (SIP

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

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

SIP Reply Message Status codeand function Type Status Code status descriptionTemporary response (1XX) trying is in process180 ringing ringing181 Call being forwarder calls are forwardLine queue181* Session Progress Sessions Session succeeded (2XX) OK session succeeded Redirect (3XX) multiple multiple selectionMoved Permanently permanent move302 moved temporaily temporary movement305 Use Proxy User agent380 Alternative Service Alternative services Req

1xx = notification response 100 trying 180 dialing in progress 181 being transferred 182 queuing 183 call progress 2XX = successful response 200 OK 202 accepted: used for referral 3xx = Transfer Response Over 300 options 301 permanent migration 302 temporarily migrated 305 use Proxy Server 380 alternative services 4xx = call failed 400 improper call 401 unauthorized: only for use by the Registry. The proxy server should use the proxy server for authorization 407 402 payment

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

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

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

This article original from the http://blog.csdn.net/voipmaker reprint indicate the source. Dual-stream is the concept in video conferencing. It generally means that the client can display two video streams at the same time, one is the main video (main), and the other is usually the content sharing (slides ), it is also called share content. Content is usually screen, PPT, document, and other content. The SIP protocol implements dual-stream. The

SIP is a tool for generating C + + interface code for Python, similar to SWIG, but using a different interface format. The idea originated in Swig, primarily to create the QT package for Python, which was used to create PyQt and Pykde , and to support the QT Signal/slot system. This article mainly introduces the compilation and installation of SIP and the birth of C + + code into Python under the window pla

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

The SIP Protocol provides a standard-based IP communication method for multiple devices and applications. This White Paper describes the support for the SIP protocol in the Cisco communications system. Cisco's Unified Communication System includes IP voice, data and video communication products and applications, which can help organizations communicate more effectively, simplify business processes, and achi

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 establ

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

"swatchmate cube" in smart devices that appeared in 2014 was definitely a surprise to me. As a portable color detection device, it can capture the colors from any object surface, helping users to easily capture these colors. If the cube is reliable, the designer will be able to capture all the wonderful colours of nature through it. So it has become one of the first devices I want to start this year. But the cube is not listed, obviously it is not the protagonist I want to introduce

In the process of building PYQT I met a very disgusting problem, in the installation of the SIP after compiling the source code after the installation process has been prompted me: Operation not permitted , I even reinstall the system is useless, finally through the data to overcome the problem. Installing SIPDownload the SIP source code package after extracting it into its folder:python configure

I. SIP Concept Session Initiation Protocol (SIP) is an application-layer control Protocol used to establish, modify, and terminate multimedia sessions. sessions can be IP phones, multimedia distribution, and multimedia meetings. It is the core protocol of the IETF multimedia data and control architecture (RFC3261 is the latest RFC document ). The main purpose is to solve the signaling control in the IP netw