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During our understanding of the SIP protocol stack, we found that the application of the SIP Protocol can be implemented on many platforms and devices. The following software is required for the establishment of the SIP Soft Phone development environment to develop a
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 applic
Translated from:Http://developer.android.com/guide/topics/connectivity/sip.html Android provides APIs that support the Session Initiation Protocol (SIP. This allows you to add the SIP-based Internet phone function to your application.Program. Android contains a complete SIP proto
In The XML configuration file of jain sip proxy, The proxy needs to initialize it through The XML file. Therefore, we need to know a lot about this part of content. Let's take a look at the parameters of the SIP protocol stack you configured. So we will give a detailed explanation of this Part, and hope it will help you. SIP_STACK tag: this parameter is required.
Session Initiation Protocol (SIP) is the biggest winner in the VoiceCon IP Phone exhibition held this week. Some vendors have announced the launch of SIP-based upgrade products or added support for this security protocol to their existing products.SIP is a single signaling and Event Notification
... 13 8.6.2.1 when the communication parties are at different levels of Nat... 14 8.6.2.2 related to the NAT type... 15 8.6.2.3 other cases... 16 8.6.2.4 peer reflexive in Internet p2p... 16 9 Application of ice in SIP... 16 9.1 both parties collect three groups of addresses ...... 17 9.1 A sends invite to B... 18 9.2 B returns 100, 101, 180 to a... 18 9.3 B returns 200 OK to a... 19 9.4 A returns ack to B... 19 Last week I wrote 1st, 2, 3, 4, and 5
In the previous article, we analyzed the process of registering a message with a sip service. Next we will analyze the process of processing an invite request. From the handle_request_invite entry, the invite request processes the replace Request Header here. If it is replace, it is considered to be consulting. At this time, no new channel is created, but a channel is found (masqued ), in most cases, a new request is created based on invite. Therefor
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
1 Description This article describes in detail the P2P SIP telephone process based on the STUN series protocol, which involves the interaction of SIP signaling, the principles of P2P, and Protocol interaction of STUN, TURN, and ICE. The interaction between service units mentioned in this article uses UDP, which does no
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
* 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
Reprint: http://www.cnblogs.com/ishangs/p/3816689.htmlApplication of Stun/turn/ice protocol in peer-to sip (II.)1 description2 dozen holes and the concept of crossing ... 13 hitting holes and crossing ... 24 using the STUN Series Protocol traversal features ... 25 Stun/turn/ice The relationship of the agreement ... 36 Stun pr
the RFC, as its language is rather clear. I shoshould mention that there is a new draft updating RFC 3265, which addresses when issues that have come up in recent years. some changes clarify the text, others alter some definitions (e.g ., the dialog is now created only when the specified y Transaction Completes ). other changes also discourage multiple usages on a single dialog.FinalePart 1 focused on the SIP foundations and showed the Protocol's sim
Tags: style blog http OS strong ar art Div log The SIP protocol is used in the national standard of the security video system. This document describes and develops a set of SIP protocol components. The exosip2 and osip2 libraries are generally used when developing such systems. This is an open-source
streams have a public media format 415 Response ( Media type not supported ) , and join 304 Warning Header field ( Media type not supported ) . 3 . Multicast Operations( 1 ) The multicast addresses that are accepted and sent are the same. ( 2 ) is called not allowed to change the media stream only hair, just accept or receive / To the hair characteristics. ( 3 ) If the call does not support multicasting, the loopback - Response and the Warning ( multicast not available ) . 4 . delayed Media
8.1.3.5. 4.14 416 Unsupported URI Scheme The server terminated processing the request because it did not support the URI scheme in Request-uri. The client handles this reply reference 8.1.3.5. 4.15 Bad Extension The server does not know the protocol extension as indicated in the Proxy-require (20.29) or Require (20.32) header field in the request. The server must list unsupported extensions in the unsupported header field. UAC handles this response s
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 prot
Csdn lidp http://blog.csdn.net/perfectpdl The SIP response to the invite request may be final or temporary. The final response is always sent reliably, but not the temporary response. You can use the prack (temporary response confirmation) method to reliably send a temporary response.To develop applications that support prack, the following conditions must be met: The client sending the invite request must put a 100rel tag in the supported or requ
SDP application in the SIP protocol and SDPSIP Application The SDP is used to construct the message bodies of INVITE, 200OK, and ACK messages for the master and called users to exchange media information. 1. Media Stream Configuration (1) The description of the primary called media must correspond to the nth media stream (m =) of the primary called, and both contain a = rtpmap. this aims to adapt to the con
============ Problem Description ============Not involved in audio, video send, as long as the implementation of registration, and chat function on the line, the online sipdroid source, but the configuration of the XML ============ Solution 1============9 is the Android 2.3 version, it should be very few machines are less than 2.3 of the bar, so this program can be installedAndroid platform based on the SIP protoc
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