pldt sip

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

P2p-sip is a peer-to-peer telephone protocol, and someone wrote a python implementation. This only supports python2,2.6 above PIP installation, or download installation package decompression. After decompression has the readme, chews the English. Write webcaller.py Import gevent, sys from gevent import monkey; Monkey.patch_all () from GEVENT.PYWSGI import wsgiserver to CGI import Parse_qs, escape import logging from logging Impor T config logging.co

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

SIP Reply Message Status codeand featureType Status Code status descriptionTemporary response (1XX) Trying is in processRinging ringing181 call being forwarder is forward182 Queue Queue181* Session Progress SessionsSession success (2XX) OK session succeededRedirect (3XX) multiple multiple options301 Moved Permanently permanent mobile302 moved temporaily temporary movement305 Use Proxy User agent380 Alternative service Replacement services Request fail

650) this.width=650; "title=" clip_image002 "style=" border-top:0px;border-right:0px;background-image:none; border-bottom:0px;padding-top:0px;padding-left:0px;margin:0px;border-left:0px;padding-right:0px; "border=" 0 "alt = "clip_image002" src= "Http://s3.51cto.com/wyfs02/M00/84/57/wKiom1eNzEnCj0QxAABdCgY5914328.gif" height= "384"/>The difference between H323 and sipSIP P2p:trunkSIP C/S: End pointSIP dialing behavior does not support KPML. Every keystroke is sent once.The default

Background: Generally, when a UA receives a request that creates a dialog (invite/Subscribe/refer), it must decide whether to authorize the request, in some cases, UA determines whether to authenticate the request by determining whether the request is in a created dialog, For example, after invite creates a dialog, UA does not need to authenticate other requests (prack, Act, etc.) sent in this dialog, but the problem is that for refer, message, and SUBSCRIBE requests, A new dialog will be crea

Learn some of the FreeSWITCH core commands, and then learn more about FS in detail.To see if it was not previously suspected, two times when programming changes the configuration file, or Java injects some parameters into the configuration file, learn more about the following configuration file.This should be difficult, not clear.Ask Mr. Baidu.Learn a new knowledge of how FS adds recording functions to configureThe general telephone system can record voice calls in the system, and voice recordin

In sip, both re-invite and update are used to modify the session parameter. The difference is that update does not affect the status of the dialog, and re-invite changes the status of the dialog. Therefore, update can be sent before the first invite is responded (that is, before 200ok of invite is received ). That is to say, update can be used to control early media. The re-invite can only be sent after the first invite cup responds (that is, after th

InProgramWhen the MessageBox pop-up error message is applied, when you click the OK button, it is found that the keyboard icon on the mainmenu has disappeared. You must click it once to appear. Although it does not affect the use of normal functions, it may be confusing for PDA cainiao, and many methods to refresh the interface cannot be solved. Baidu cannot find the answer to this strange problem. You can directly query the problem on Google and find that foreigners also encounter this proble