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[Android Application Development]-(23) Android SIP support

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

Android Internet telephony development, VOIP/SIP so many open source choose which good

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

Specification for JAIN-SIP Requirement

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, iotsip

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 in the SIP protocol

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

Application of SDP in SIP protocol

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

How to: TCPDUMP sip VoIP capture FreeBSD tutorial

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

Pyhotn's P2P-SIP network phone test

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

SIP protocol overview audio and video tutorial

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

Asterisk 1.8 SIP protocol stack Analysis 2

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

Research results of SIP audio calls in Android

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

Cisco SIP VoIP architecture solution

The Cisco SIP VoIP architecture solution provides users with many services. The table lists various IP telephone services that can be implemented by Cisco SIP VoIP Architecture solutions.Cisco SIP VoIP architecture solution ComponentsThe Cisco SIP VoIP architecture solution consists of the following elements:Sip ip tel

Rtp sip configuration details

The SIP protocol and some of its applications are worth learning. In this regard, we will explain the configuration of rtp sip today. SIP (Session Initiation Protocol) is usually used for VOIP calls, call establishment, call negotiation, and call termination. it helps two terminals to recognize each other, but it does not process media. When A call is established

VoIP in-depth: An Introduction to the SIP protocol, part 2, 3-4

former scenario usually involves early media, such as Ring-Back Tone (the music you hear when calling a person subscribed to this service) or interworking with PSTN. the later scenario may involve resource reservation. these scenarios, by definition, require setting and changing the media properties as the call begins, and this forces sip to take a complex path to support them. The following flow demonstrates a complex invite scenario. For clarity pu

Receiving __freeswitch of FreeSWITCH SIP signaling

Processing of FreeSWITCH SIP signaling in Mod_sofiaFirst, a thread that handles SIP messages is created inside the module's load (mod_sofia_load): /* Start one message thread /switch_log_printf (Switch_channel_log, Switch_log_info, "starting initial message Thread.\n "); Sofia_msg_thread_start (0); The Config_sofia function is then called. if (Config_sofia (Sofia_config_load, NULL)!= switch

Asterisk SIP MySQL Configuration

Label:Recently studied asterisk configuration, before the SIP account is configured in the sip.conf file, manual writing dead, the current demand, is the dynamic new SIP account, saved in the database.After adding data to the database, use the command SIP show users to not load the SIP account to the database.1. Downlo

Sip rfc 4538 authorization request through dialog

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

The next day, learn about SIP (FreeSWITCH add recording function) (2)

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

Differences between update and re-invite methods in SIP

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

Conflict between MessageBox and sip on PDA Platform

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

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