sip sdp

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Generate SDP file for DSS

Here we will introduce two methods to generate SDP files for your reference only. 1Environment Operating System:Centos6.2 _ 64 Kernel version:2.6.32-220.23.1.el6.x86 _ 64 Darwin Streaming ServerVersion:6.0.3 Mpeg4ipVersion:1.6.1 FFmpegVersion:0.6.5 2,UseMpeg4ipGenerateSDPFile Easy to installMpeg4ipDirectly runMp4liveYou can, for example: 3, UseFFmpegGenerateSDPFile The command is as follows: FFmpeg-F video4linux2-s 176*144-R 10-

Summary of the advantages of SIP

The use of the SIP protocol is not uncommon. With the popularity of the VoIP service, we also have a better understanding of the SIP protocol. Here we will focus on the advantages of SIP. In this case, we need to compare this advantage with other protocols. Advantages of SIP: Service providers can select standard compo

Java 7 SDP: Once written, running everywhere, sometimes also run super dazzle!

This article will briefly introduce the Java Socket Direct protocol (Sockets Direct PROTOCOL,SDP) introduced in the Java 7 SDK, and this new technology is a very exciting breakthrough. If you want to native access to InfiniBand's remote Direct memory access (Sqlremote direct Memory ACCESS,RDMA), SDP enables ultra-high performance computing (Ultra high Performance Computing, UHPC) Community uses Java-generic

Bluetooth SDP Introduction

SDP, service Discovery Protocol, Services Discovery Protocol 1. Concept The SDP provides a way to discover services and the properties of those available services, but it does not provide a mechanism to take advantage of these services. The schema is Client-server mode, as shown in the following figure SDP Server maintains a list of service records, with each en

Comparison between BICC and SIP-NGN Protocol

supports the bearer of call control signals on MTP SS7 and ATM networks, and CS2 supports the bearer over IP networks, CS3 focuses on the quality of bearer applications such as MPLS and IP QoS and the intercommunication with SIP. Many equipment manufacturers and operators have participated in the formulation of CS3 standards. (2) SIP-T Before introducing the SIP

Research on ice-based sip signaling penetration over symmetric NAT technology

, ice is a good fit for such requirements..2 ice technology 2.1 ice Introduction Interactive connectivity establishment method (ICE) is not a new protocol,It does not need to expand stun, turn, or rsip to apply to various Nat services. Ice is a comprehensive application of the aboveAgreement to make it work in the most appropriate circumstances, to make up for the inherent defects arising from the independent use of any of them. ForFor sip, ice only

Introduction to open-source VoIP Based on the SIP and RTP protocols-qutecom

behalf of other clients, acting as both a server and a client. Before forwarding a request, it can rewrite the content in the original request message. Redirect server: it receives the SIP request and maps the original address in the request to zero or multiple new addresses and returns them to the client. Registration server: it receives client registration requests to complete user address registration. Using

A Preliminary Study of IMS service provision Method Based on SIP Application Server

communication, IMS is attracting more and more attention. As one of the important members of the TD-SCDMA Industry Alliance, Putian Information Technology Research Institute is actively researching and developing IMS systems, and will finally provide complete IMS system solutions to operators. References [1] miikka unzip kselka, Georg Mayer, Hisham, kharbilirubin, Aki Niemi. The IMS: IP Multimedia concepts and services in the Mobile Domain, June, 2004[2] 3GPP ts 23.228: "IP Multimedia Subsystem

SIP Protocol Document Translation

not depend on any of these protocols.SIP does not provide services. Of course, SIP provides primitives for implementing different services. For example, SIP can locate a user and transfer a non-transparent object to his current location. If this primitive is used to transmit the session description is written to the SDP (Sessiong Description Protocol), for examp

Research on Nat penetration Based on SIP

1 IntroductionVoIP is the most representative of the development of the new generation of Internet age.One of the application technologies. As a signaling control protocol in VoIPThere is great potential for growth. Therefore, in order to better promote the development of VoIP servicesIt will be a major research topic to solve the problem of SIP traversing NAT. This article mainlyBased on the stun method, it aims at its inability to traverse symmetric

SIP Message Type and Message format

is described using another protocol, is the Session Description Protocol SDP .) Request Message Request Line = method + space + request address + sip version number + blank line A request line is used as the starting line. The request line includes the method name, request URL, Protocol version number, and separated by spaces. Six request methods: Invite sends a call Session Request Ack invite request fina

Sip invite Process

connection instruction, this response also allows the caller to specify the format of the connected media allowed by the callee to confirm whether the caller can receive the media. The message body is as follows: SIP/2.0 200 OKVia: SIP/2.0/UDP lab.high-voltage.org: 5060; branch = z9hG4bKfw19b; Received = 100.101.102.103To: G. Marconi From: Nikola Tesla Call-ID: 123456789@lab.high-voltage.orgCSeq: 1 INVITEC

SIP (2)

status.Prack:Similar to ACK, but used for temporary response.Subscribe: This method is used to subscribe notifications of status changes to the remote endpoint.Notify:This method sends a message to notify the subscription of changes in its predefined status.Update:Allow the customer to update the parameters of a session without affecting the current state of the session.Message:The request body carries real-time messages.Refer:This function instructs the recipient to contact a third party by us

Several open source SIP stack comparison Opal,vocal,sipx,resiprocate,osip

support the newest rfc3261,resiprocate was born, but now, Resiprocate has become an independent SIP stack, it is very stable, and many commercial programs are in use.SVN: http://scm.sipfoundry.org/viewsvn/resiprocate/main/sip/Language:c++VxWorks Port:noWin32 Port:yesLinux Port:yesSupports RFC 3261:yesSupports RFC 2327:yesSupports RFC 3264:yesSupports RFC 3263:partialSupports RFC 3515:yesSupports RFC 3262:n

SIP protocol parsing and implementation (C and C ++ use Osip) 11

/1.1. This behavior can be used for the "trace route (traceroute)" function to check the capabilities of individual servers in the message routing process by sending a series of ascending max-forwards value options requests. As a general UA behavior, if options does not respond for a long time, the transaction layer can return a timeout error. This indicates that the target is inaccessible and the query capability is unavailable. The options request may be sent by one end of the created dialog t

Reliability of temporary response in SIP protocol

@bell-tel.com;tag=11 Call-id:70710@saturn.bell-tel.com Cseq:1 INVITE CONTENT-TYPE:APPLICATION/SDP v=0 S=let ' s talk b=ct:128 C=in IP4 north.east.isi.edu M=audio 3456 RTP/AVP 5 0 7 m=video 2232 RTP/AVP 31 C->s:prack sip:watson@bell-tel.com sip/2.0 rack:776655 1 INVITE VIA:SIP/2.0/UDP saturn.bell-tel.com From:sip:alexander@bell-tel.com to:sip:watson@bell-tel.com;tag=11 Call-id:70710@saturn.bell-tel.com

SIP Proxy Server PartySIP and osippartysip Based on oSIP open source Library

the same as that of the eXosip framework, except that the functions of the modules are different. Ii. Official website of PartySIP Proxy Server The official website of the PartySIP proxy server is shown in: Open source Web site: http://www.nongnu.org/partysip/partysip.html Iii. Introduction to oSIP open source Library OSIP is small and flexible. It is compiled based on standard C and can be used in any POSIX-supporting system. Therefore, it can be widely used in embedded systems. The main fe

Learning notes-SIP message receiving and processing flow based on sipdroid SIP audio and video telephony

/success (actual Sipprovider callback), the Fail/sucess interface of the transactionclient is called Public classTransactionclientextendsTransaction {Transactionclientlistener transaction_listener; Publictransactionclient (Sipprovider sip_provider, Message req, Transactionclientlistener listener) {Super(Sip_provider); Request=NewMessage (req); Init (Listener, Request.gettransactionid ()); //This.transaction_listener = listener;} //The actual Sipprovider callback Public voidOnreceivedmessage (Si

Next-generation Internet protocols led by WLCP of SIP--BEA Systems (1)

applications;Completely part of the service-oriented architecture SOA)Can be displayed in any type of browser or device, also known as multi-channel transmission;Provides a personalized view of applications through the WebLogic Portal.At present, the capabilities of these combinations are generally referred to as the Service Delivery Platform, SDP ). This example blur the differences between wireless, wired, VoIP, and Web-based transfer channels. In

Google released the world's first open-source HTML5 SIP client

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

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