sip speakerphone

IntroducedThis article is a reference guide for SIP4.18. SIP is a python tool used to automatically generate Python bindings to C/C + + libraries. SIP was originally developed in 1998 with PYQT for Python and QT GUI Toolkit, but is suitable for generating bindings for any C or C + + libraries.This version of the SIP-generated bindings is available for Python vers

Kamailio is an open-source SIP server, formerly known as openser. Kamailio is an open source, gpl2, SIP Server Routing platform. It is written in C for Linux/Unix plaforms and focuses on performance, flexibility and security. On Nov 04,200 8, kamailio and SIP Express Router have started the SIP router project. Web Link

BICC vs sip-Comparison of NGN Protocol -------------------------------------------------------------------------------- One notable feature of the Next Generation Network Based on Softswitch is the separation of Call Control and bearer control. In terms of call and control of communication, from the Standard Research of the International Telecommunication Union (ITU-T) and Internet Engineering Task Group (IETF), there are two protocols that deserve

client should try again in the sip uri mode. If you receive a 420 (incorrect extension) Response (rfc3261 section 21.4.15 ), the request lists an extension option not supported by the proxy or uas in the require or proxy-require header domain. UAC should ignore the extensions listed in the response's unsupported header and try again. In all the above cases, the request is modified accordingly and a new request is created. This new request creates a n

Simple Solution: c: \ Python26 \ python.exe setup. py py2exe -- extends des sip It is very convenient to use pyqt to complete the form interface, but there will be problems after packaging it into exe. The solution on the internet is as follows:Another Solution to the same problem: from distutils.core import setupimport py2exesetup(windows=[{"script":"main.py"}], options={"py2exe":{"includes":["sip"]}})

According to the RFC3261-13.2.1, the offer/answer model used by the SIP is established in the dialog environment. The RFC also specifically imposes restrictions on offer/answer Interaction: 1.The initial offer must be in the invite message or the first reliable non-Failed response. Note: At that time, the reliability Effect of rfc3261 was only 2 **. Next we will talk about 1 ** (except 100. 2.If the initial offer is in the invite message, the answer m

Status Codes and type status codes of SIP response messagesTemporary response (1xx) 100 trying in process180 ringing181 call being forwarder call forward182 queue181 * session progress session Successful SESSION (2XX) 200 OK session successfulRedirection (3xx) 300 multiple multi-Choice301 moved permanently permanent movement302 moved temporaily temporary movement305 use proxy user proxy380 alternative serviceRequest failed (4xx) 400 bad request error

Abstract: IP MultimediaSubsystem ( IMSThe user plane, control plane, and business plane are separated to facilitate the provision of services to users. This article will focus on analysis SIPApplication Server(SIP as) Issues related to the IMS service provision process, including the general process, billing functions at the business layer, and technical difficulties provided by the Service, analytics is based on a specific service (Message Service) p

I created a freeswitch kernel research and exchange group,45211986. welcome to join us. In addition, we accept the SIP-based communication server and client solutions. Freeswitch supports cross-platform, and the underlying library uses Apache Portable Runtime lib (APR). The author has been familiar with Apache in his previous work.In swith_apr.c, all the tool library interface functions provided by APR are referenced by freeswitch for the purpose

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

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 protocol stack and integrates the call management service, so that applications can easi

RADVISION's SIP server platform is a framework for building all SIP servers and supporting fast and effective application software development for SIP servers, as RADVISION's SIP toolkit is in the leading position in the market, therefore, the platform implements the functions of the

I believe everyone is in touch with the SIP protocol. In the previous article, we also discussed the content of the SIP protocol. If you are not clear about it, you can review it a little. We know that the SIP protocol has a huge development potential in the network communication field. Here we will explain the content of the IMS and

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