sip sdp

the calling application, used to indicate that the call has received the notification. Route Set: Route Set. A route set is a set of ordered Sip/SIPs Uris, that is, the list of proxy servers that pass through when a special request is sent. A route set can be learned by using a field similar to the record-Route Header or configured. Server: Server. A server is a network entity that receives requests. It provides services based on the requests and ret

: Which means something else entirely. the '; expires = 1800' inside the angle brackets is a part of the URI, not a field parameter of the contact field. the '; expires = 2970' outside is the field parameter, so it is correct to use itThe expiration time of the contact. The inner one is used by the proxy to control forking timeouts when this contact is used, and has nothing to do with the Registry expiration. Broad Band 1.2. Basic Process 1.2.1. Main Process The

Video courses and related documents code address: https://github.com/EasyDarwin/Course#course-3RTP ProtocolThe real-time Transport protocol RTP (real-time Transport Protocol) is a network transport protocol that was published by the IETF Multimedia Transmission Working Group in RFC 1889 in 1996 and later updated in RFC3550.ITU-T also released its own RTP documentation as a h.225.0, but it was later canceled when the IETF released a standard RFC on its stability. It is described in detail as an I

SDP fileIntroductionThe session Description Protocol (SDP) is a format for describing the initialization parameters of streaming media Session S. SDP does not deliver media itself are used for negotiation between end points of media type, format, and all Associa Ted Properties.To connect to a streaming server sending an MPEG2 transport stream, you don't have usua

Keywords: Bluetooth bluetooth® protocol SDP Anatomy SDP overview SDP notes Author: Zhongjun In the spirit of mutual learning purposes, to share this series of articles, welcome reprint, please specify the author, respect for copyright, thank you Please correct me if the article is not in the wrong place, learn together You can also access my csdn:http

Analysis of SIP route and record_route/SIP Routing Mechanism from: [url] Routing Analysis of SIP Routing Mechanism (zz)We have already introduced sip-related knowledge about important SIP header domains, registration processes, and session processes. Now we will introduce t

In fact, SDP is a very simple protocol. The key is to understand the concept. To be honest, the concept here is indeed very messy. Service: A service is an entity of a service class. used to provide information and execute an action. it can be composed of software, hardware, or a combination of the two. the Service handle attribute represents its key attribute. Service record: stores information about a service. A service record is composed of a servi

phones as well-known traditional phones, but the transmission mode has changed from circuit switching to group switching. The SIP Protocol focuses on using IP phones as an application on the Internet, which has higher requirements for signaling and QoS than FTP and E-mail, they support basically the same business, and also use RTP as the media transmission protocol. But H.323 is a relatively complex protocol.H.323 uses a binary method based on ASN.1

Original link http://www.cnblogs.com/yoyotl/p/5649648.html If there is infringement, please contact Delete, thank you for sharing. First, the SDP specification of the format of the reply description, generally combined with the session protocol work together. Common session transfer protocols include: SAP (sessionannouncement Protocol Session Announcement Protocol), Sip,rtsp,http, and e-mail using MIME. (P

----------------------------------------------------------------------------RFC list:3550 (RTP/RTCP)-3711 (SRTP)-5245 (ICE)-3261 (SIP)-4575 (sip-conference)-4566 (SDP)5389 (Stun)-5766 (Turn)-6455 (WebSocket)-6865 (FEC)-2616 (HTTP)----------------------------------------------------------------------------One, RFC 5245 ICE Study Notes------------------------------

(1)(2)(3)-------------AUTHOR:PKF------------------time:2015-1-6Later discovered is that the server address of the connection is specified so that it can not play, and then with Wireshark capture a lot of UDP data came out, is not a port problem, and RTSP head problemIf it is an RTSP contract, the packet header will have a 4-byte symbol for a specific frame:Framingheader[0] = ' $ ';FRAMINGHEADER[1] = Streamchannelid;FRAMINGHEADER[2] = (u_int8_t) ((packetsize0xff00) >>8);FRAMINGHEADER[3] = (u_int8

This tool is a real-time analysis of the SIP Communication Protocol, which is the main function of the Communication Engineering of zhongzheng University. In addition, the distributed design can be used to analyze cross-domain sip packets, draw a complete signaling flowchart, and control the web interface... background in view of the fact that there is no free communication protocol analysis software for th

helps the devices verify that the network is actually transferring all the packets and that the quality is acceptable. There are several ways to calculate quality; most of these methods compare the time packets arrived to the time they shocould have arrived, and they also detect and count the number of packets that were lost.VoIP elementsoffer-answer The default media negotiation protocol is the Session Description Protocol or SDP. this is defined by

prevent cyclic jumping= per proxy server, the integer minus oneTo:g. Marconi From:nikola Tesla = Indicates the sender and target party of the request message= If there is a user name tag in it, the address needs to be wrapped in angle brackets.= for INVITE messages, you can include tag in the from field , which is also a random codeCall-id: [email protected]The Identifier (dialog indentifier) on both sides of the call.Cseq:1 INVITEcommand seqence, each time a new request is sent, the number is

parameters; Call Establishment: the establishment of the session parameters of the caller and the called party; Call Management: includes transferring and terminating sessions, modifying call parameters, and calling services. The SIP protocol can be combined with other IETF protocols to establish a sound multimedia structure, such as providing real-time data transmission and service quality QOS) Feedback of real-time transmission protocol RTP) real-t

and Authorization header fields of the SIP protocol. when the UE sends a registration or call request to CSCF, it must provide security parameters such as the Protocol identity and password in the Authorization header of the REGISTER message. When the UE does not contain security parameters, CSCF will send a 401 response (unauthorized) to the UE, including the WWW-authenticate field. The WWW-authenticate field carries the necessary security parameter

gateway or proxy server. The following problems need to be solved through DNS:1. E.164 address ing with the SIP URL2. Address ing between E.164 and MG endpoint identity3. Address ing between the sip url and the MG endpoint identityAll three methods can be implemented through database matching. The most important and difficult of these methods is the ing between E.164 and

Via:sip/2.0/udp lab.high-voltage.org:5060;branch=z9hg4bkfw19b = SIP version number ( 2.0 Udp), call address, = Branch transport identity = Via = transport type can be udp tcp tls, SCTP Max-forwards:70= Maximum number of hops, is the number of hops through the SIP server, mainly to prevent cyclic jumping= per proxy server, the integer minus oneTo:g. MarconiFrom:nikola Tesla; tag=76341

The previous SDP did not parse the "A = fmtp" field for the h264 encoded video. Today, it is added to parse the width and height of the video. In section 8.2 of rfc3984, field is introduced. Here we only decode the sprop-parameter-sets field, because my main purpose is to parse the video width and height information. This field is encoded with base-64. Therefore, the base-64 is decoded first. The decoding method is provided here: long CBase64::Dec

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