polycom sip

Developed for SIP/IMS video clients, supports access to sip Softswitch, IMS core network, andVoice, video, and instant messaging functions. The video formats support h263, h264, and MPEG4 soft encoding solutions. The hardware coding/decoding interface is provided for interconnection and servers. If you are interested, contact me. Implements standard-based (SIP

At the request of the dynaguy brothers, I have attached a relatively complete experiment to colleagues who are still exploring: (There are some problems with the 2.0beta experiment, so I will demonstrate it with the most stable 1.2.3) In this test, we did not discuss the issue of using the SIP Trunk directly to connect other sip servers without board installation to enable the

In the process of building PYQT I met a very disgusting problem, in the installation of SIP after compiling the source of the installation process has been prompted me: Operation not permitted , I even reinstall the system is useless, and finally through the data to solve the problem. Installing SIPDownload the SIP source package after extracting it into its directory:python configure.pysudo makesudo m

SIP Redirect The SIP Redirect feature allows the IMG to respond to the 3xx class of SIP messages returned from a Redirect server. The 3XX responses provide information about a user ' s new location, or alternative services that is able to satisfy the Call. This feature was based on RFCs 3261 section 8.1.3.4 and RFC 2543. In a

Release date:Updated on: Affected Systems:Yealink Yealink SIP-T20P IP Phone Description:--------------------------------------------------------------------------------Bugtraq id: 57029Yealink SIP-T20P is an IP Phone.YeaLink IP Phone SIP-TxxP The vulnerability is described as follows:1) The default username ("user") and password ("user") can access the hidden pa

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 not involve TCP holes and other TCP-related operations. This document assumes that neither p

Both SIP and XMPP are application-layer protocols that are used primarily to send voice and instant messaging over the Internet im,rfc3521 defines the sip,rfc3920 definition of XMPP. XMPP comes from instant messaging systems, SIP-like voice and video communications.XMPP protocol is mainly responsible for the exchange of data,

Is the call flowchart of Asterisk: We use the call process of SIP as an example to describe the call process of other channels. The call process (incoming) is as follows: Do_monitor-> sipsock_read-> handle_request-> handle_request_invite-> sip_new/ast_pbx_start-> pbx_thread->__ ast_pbx_run -> Ast_spawn_extension-> pbx_extension_helper-> pbx_exec-> execute dialplan When the chan_sip module is loaded, an independent listening thread do_monitor is starte

Section 3 redirect servers In some frameworks, relying on the proxy server can reduce the load on the proxy server, which is beneficial for forwarding requests and enhancing signals. Redirection allows the server to send route information to the client through the response to the request, so it frees itself from the subsequent message loop of the transaction, at the same time, it can continue to accurately locate the request target. When the request's original sender receives a redirection, It r

Chapter 9 dialogue A key concept for user proxy is dialog. A dialog indicates a point-to-point sip connection between two user proxies at some time. The dialog ensures that messages between user proxies are ordered and correctly routed. A dialog indicates the context of a SIP message. Rfc3261 the UA processing discussed in section 8th is irrelevant to the method. This chapter discusses how to construct a di

Haha, if you have never touched network programming, don't look down. Give a definition first: SIP (Session Initial Protocol) is a signaling protocol that is used to set up, modify and terminate sessions, like Internet phone calland multimedia conferences between two participant ants. For Translation: SIP is a signaling protocol used to establish, modify, and terminate sessions, such as network call

1xx = notification response 100 trying 180 dialing in progress 181 being transferred 182 queuing 183 call progress 2XX = successful response 200 OK 202 accepted: used for referral 3xx = Transfer Response Over 300 options 301 permanent migration 302 temporarily migrated 305 use Proxy Server 380 alternative services 4xx = call failed 400 improper call 401 unauthorized: only for use by the Registry. The proxy server should use the proxy server for authorization 407 402 payment

Use FireBreath to develop real-time playback interfaces (Yate + SIP + FFMPEG + SDL) and firebreathyate At that time, such a blog post was really needed to guide this function module. Unfortunately, FireBreath has very little information on the Internet and is not very familiar with C ++, so we tried and explored it all the way. Fortunately, we have implemented this module, and now we have recorded it. First of all, our Yate

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 SIP protocol stack library. The actual requirements cannot be

I began to study the VOIP/SIP agreement from 09, open source project also saw a few, the earliest Pjsip 05 began to push the time, began to pay attention to, also in their own winmobile project used. Later also saw Sipdroid,imsdroid (Doubango), Linphone,csipsimple (PJSIP).I think the best advantage of Linphone and Csipsimple,linphone is the full platform support, Android,ios,winphone,windows,linux,mac osx,web all support, but the quality is still unde

Requirement Specification Directory 1: Target and scope 1.1jain-Sip API description 2: Introduction 2.1 System Overview 3: requirement definition 3.1 jainsip Target Model 3.2 jainsip naming conventions 3.3 jainsip Structure 3.4 General primitives 4. External requirements 4.1 External Interface 4.2 Resource requirements 4.3 acceptance test release 4.4 documentation requirements 4.5 lightweight requirements 4.6 Quality Requirements 4.7

IOT command (based on sip) client API design for java, iotsipThe Iot Device Control we implement is implemented by extending the sip protocol. Because pjsip is implemented based on pjsip, while pjsip uses C Programming, how to make the business layer (android end, java) easier to use the provided command API is the focus, the original method is to encapsulate (C ---> jni ---> java) from the underlying c, wh

Strict routing and loose Routing 1. The address list of a loose route does not list a complete and strict path, but only provides some key points in the path..You can use the automatic route selection function of the vro to route data between key points. data packets must also be copied during data packet sharding. In a SIP message, if the parameter in the first Route Header field contains the LR parameter, It is a loose route. 2. Strict routing re

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

I recently found out doing a sip capture on a FreeBSD system is a little different than centos or other Linux distributions. here is a easy to use command that will grab the SIP packets from TCP dump. this will give you an easy to read text file for debugging or tracing. > Tcpdump-I bce1-n-S0-vvv UDP port 5060>/usr/src/capture_file Let's go over the options for this command: -I = interface which on my BS