sip ss7

SIP (Session Initiation Protocol, conversational initial protocol) was developed to help provide advanced telephony services across the Internet. Internet telephony (IP telephony) is evolving into a formal business telephony model, and sip is an important part of the NGN (Next Generation Network) family of protocols needed to ensure this evolution. Support for the H + protocolSIP Learning Note 2007-12-28 21

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

Tags: CME cucm Cisco UnifiedThere is a logical association between the commands and must be configured in a sequential order.1. Configure voice encoding and SIP address formatVoice class URI 2 sipHost IPv4:address of CUCMVoice Class codec 1Codec Preference 1 G729R8Codec Preference 2 G711ulawCodec Preference 3 G711alawCodec Preference 4 g722-642. Configure VoIP Service Specific parametersVoice service VoIPIP Address Trusted ListIPv4 The address of the

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

**************************************** **************************************** **************************************** ***Easywave time: 2014.09.14 Category: Linux application-open source server opensips introduction statement: reprinted, please keep the link NOTE: If any error occurs, please correct it. These are my Learning Log articles ...... **************************************** **************************************** **************************************** *** I. Introduction to op

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 protocol that combines voice, video, and message communication. This Protocol also provides a standard-based appr

1. What is sip? Step-by-Step sip Ii. How to Use sip in Android? Sipdroid, Demo: Android-SDK-Windows \ samples \ Android-9 \ sipdemo 3. Does the system support sip? Android provides the SIP function since 2.3. The SIP-related APIs

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

The integration of unified communication involves software, hardware, mobile devices, and fixed devices. The standards and protocols are different. Therefore, the integration is difficult. In terms of IP phones and desktop applications, there are mainly SIP and H.323 Protocols; mobile videos are H.324M; Monitoring also includes H.323, SIP and proprietary protocols as well as MEPG2, MEPG4, and H. 264 and ot

SIP is an application control signaling protocol proposed by IETF. As the name implies, it is used to initiate a session. It can be used to create, modify, and end multimedia session processes attended by multiple participants. Participants can communicate with each other through multicast, unicast, or network connection. There are clients and servers in the SIP. A client is an application that establishes

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

First, we will briefly introduce the SIP protocol, which is a Session Initiation Protocol and mainly used for network multimedia calls. The sip api can be called only in android2.3 or later versions, and the device must support the sip before making a SIP call. The APIs used by SIP