voip

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Comparison of open-source Android Voip clients

On the Android platform, the open-source projects of the Voip client include Sipdroid, IMSdroid, CSipSimple, and Linphone. The following compares them: Link \ Client Sipdroid IMSdroid CSipSimple Linphone Sip 3GPP IMS Pjsip Osip Tool Language Java (Architecture) C ++ (encoding) Java (UI) C ++ (architecture, encoding) Java (UI) C ++ (architecture, encoding) Java C ++ Audio/Video

Brief Analysis on whether Wi-Fi can carry VoIP?

For wireless data networks, voice is a "killer application ". The high-performance Wi-Fi mesh network system is a killer IP wireless network. However, not all mesh networks are identical. As wireless mesh networks become increasingly popular-almost every day, people announce the newly deployed public and private networks. To add voice applications to business needs, the network needs to improve overall performance so as to process real-time applications. Once Multiple Relay segments (hops) appea

How to solve the security problem of Wi-Fi network VoIP (1)

Over the past few years, the penetration rate of VoIP systems in both the enterprise and residential markets has been greatly improved. VoIP integrates data and voice services into a unified network, which can greatly save costs for enterprise IT departments or home users. We also saw that various service providers began to deploy a small number of VoIP systems t

ADSL2 + gateway solution assists Aztech with IPTV and VoIP Functions

Recently, Texas Instrument (TI) announced that Aztech Systems, a global speech and data communication design and manufacturing company, will choose TI community Gateway (RG) solutions from its next-generation ADSL2 + gateway. The new Aztech product not only enriches its product series, but also enables it to provide broadband services at a low cost through the ADSL2 + network connection to achieve triple play services including IPTV and VoIP functions

Basic VOIP Configuration

Tutorial description:In this experiment, two Cisco 2600 vrouters and one Cisco 3600 vro are used to implement Voip for Division 1 and Division 2 of the blue school. The two Divisions use Cisco's 2620 as the terminal device, and the Headquarters uses Cisco's 3640 router as the center networking device. The specific structure of the connection between the leased lines is as follows:Branch 1)Cisco 2600 Voip co

Enterprise-level VoIP applications

obtain directly through their own phone:● The call forwarding service is provided unconditionally when the line is busy and there is no response;● Call waiting;● Call transfer;● Incoming call display.6. redundant bypass In addition to four telephone/fax ports, the AT-VP504E FXS also has a fifth ipvii connector for connecting to standard PSTN lines. This line is not used during normal operation. However, if the main power supply fails or the IP network goes down, the PSTN line is automatically c

H323 protocol configuration method under VoIP in checkpoint firewall

Description of the phenomenon:using the checkpoint firewall as a security gateway, the network is fine, but the Voip(H323) service is not working. Here's how to fix it:the Voip Each endpoint IP Summary Group, as the source address and destination address, see Figure a650) this.width=650; "Src=" Http://s1.51cto.com/wyfs02/M00/89/C0/wKioL1gb6rShbNPZAACyFYyb1CQ768.png-wh_500x0-wm_3 -wmp_4-s_4293603484.png "sty

VoIP technical architecture

There are some problems with the H.323 Protocol (only a limited MCU is not supported for multicast; its IP telephone network still needs to go through the Local PSTN Circuit Switching Network at the access end), and then MGCP is customized, the purpose is to break down H.323 functions into two parts: Media Gateway (MG) responsible for media processing, and the Media Gateway controller (MGC) that controls call establishment and control. Four elements of the

Voice compression and IP online voice (VoIP) technology

In the fully transparent networking scheme using IP network or leased line to realize voice switch, in addition to adopting the most advanced signaling system, the built-in voice compression platform and its IP network voice are integrated in the switching system. VOIP Technology and Gateway technology are the key to achieve high quality voice communication. The built-in speech compression technology provides a new set of solutions for the connectio

Linksys (Cisco) VoIP set audio interface-a table

Rt41p2 2fxs 4ethRt31p2 2fxs 3eth Rt3002fxs 4 + 1eth Pap2t 2fxs 1eth 10 MbpsPAP2T-NA 2fxs 1eth PAP2-NA Pap2 V2 2fxsThis product supports t38 protocol high-speed FaxIt is great for users who need to fax in China. Products prior to V2, PAP2-NA, pap2t do not support t38 Protocol(PAP2-NA, pap2t only supports SIP-based fax. There is still no way to send and receive faxes in China .)Pap2 V2, which supports t38 protocol fax, is tested by multiple carriers with a sending rate of 100%.No setup is required

Two methods for implementing IOS long Background: audiosession and VoIP

We know that IOS can get a maximum execution time of 600 seconds after enabling background tasks. How do some apps that need to be downloaded in the background or kept connected to the server exceed the limit of 600 seconds? For example, Netease open classes can be used for continuous download in the background, and Youku can also continue caching in the background. How does this happen? Generally, to enable Ios to run in the background for a long time, you need to declare

VoIP and instant messaging problems: Skype does not support IPv6

Currently, Skype can be bundled in Windows 8, but the problem is that it is not applicable to IPv6. Although there is no built-in advertisement or private space guarantee, Skype is still a speech software that is applied through Internet Protocol (VoIP) and instant messaging (IM) clients. Indeed, Microsoft is using Skype to replace other instant messaging software. However, another problem with Skype is increasingly confusing: it does not support Inte

Cloud Communications Open Platform provides converged voice, SMS, VoIP, video and IM and other communication APIs and SDKs.

Cloud Communications Open Platform provides converged voice, SMS, VoIP, video and IM and other communication APIs and SDKs. Undefined All-Star Verification-Sendcloud Undefined [Reprint] Several mainstream online development platform (PaaS) Introduction _ Purple Qin _ Sina Blog Undefined Python+selenium2+chrome building Dynamic web crawler Tools-cjsafty's Column-Blog channel-csdn.net Undefined IP Proxy API documentation, IP prox

Performance in Wan with Shunra VE SMB test system (such as video surveillance, building intercom, VOIP, IPTV, etc.)

First, Shunra VE SMB IntroductionShunra ve SMB Edition is a network simulation software product designed for small and medium sized enterprises, Shunra ve SMB Edition simulation software can be used to test, compare or predict under different network conditions-including delay, jitter, Packet loss and bandwidth (max. 10Mbps)-performance of the application or device.The software can be used to test the performance of video surveillance, building visual intercom,

A VoIP operation support system has the general SQL injection and Arbitrary File Traversal Vulnerability (a large number of enterprises are affected)

A VoIP operation support system has the general SQL Injection Arbitrary File Traversal Vulnerability (a large number of enterprises are affected) Kunshi Network Technology Co., Ltd. develops a support system for small and medium-sized scale VoIP operation services. In addition to meeting the operating rate setting and package management requirements, in addition to basic functions such as account managemen

Good news for gaming players: Install the open-source VoIP Application Mumble on Ubuntu

Good news for gaming players: Install the open-source VoIP Application Mumble on Ubuntu Mumble is a free and open-source VoIP Application released under the new BSD license. The main target user group is game players. Running is similar to TeamSpeak and Ventrilo. Users can communicate with each other by connecting to the same server. Mumble provides the following beautiful features: Low latency, which is

IP network telephony differs from VoIP network telephony

Compared to building intercom, Internet telephony can be said to have been in our lives for a short time. Generally speaking, network telephony refers to IP as the network layer protocol of the computer network voice communication system, it uses the technology collectively referred to as VoIP (Voiceover IP), that is, the use of the network to achieve voice transmission. From the technical point of view, IP network telephony is the result of integrat

Tom-skype Gold Panning personal voice did not touch VoIP policy

platform--tom Pay, Tom's game, mailbox, community fee business through it, and Tom Pay and ebay PayPal also has an interface, PayPal account can also be used to pay Tom Pay consumption. The fee paid by the user receiving the service will be divided by the Tom-skype and the service provider by 3:7. "This internet-based real-time billing voice interactive platform, currently only tom-skype." "Tom-skype is also the first partner to launch this innovative project by Skype's many regional partners

Check the network and device running status before applying VoIP.

line for one month. As you can see, there is a big difference in usage. When planning VoIP, you should assume that the traffic remains in the peak state. If this is not the case, when the processing task increases, your speech signal quality will decrease as your plan is not prepared. Another important thing is to check the cache space and discarded packets of your electronic devices. A vswitch can discard data packets. When the cache is too full, th

VoIP DTMF notes

DTMF definition: Digital keys (0 ~ 9 * # a B C D ). There are usually three methods for detecting DTMF in VoIP: SIP info, inband, and out band (rfc2833). In addition, the latest RFC has been adopted for the requirements of DTMF In the 3GPP IMS specification.4733 replaces RFC 2833. 1. Sip info For out-of-band detection, DTMF Data is transmitted through the sip signaling channel. There is no unified implementation standard, which is sent through the S

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