Codecs are used for videoconferencing and streaming media applications. Audio compressors allow to convert analog audio signals into digital ones to transmit or store them. Then the receiving device can convert the digital signals back to analog ones with the use of an audio decompressor , for playback. Lossy codecs: Lossy refers to the fact that compression results in some quality reduction. Algorithms are also used to create the impression of data being there.
Lossless codecs: These types of codecs are used to archive data in a compressed form in a way that all information of the original stream is kept. They are the proper choices if it is more important to retain the original quality than reduce data sizes especially in cases when the data is undergo further processes.
Net developers. Asked 10 years, 5 months ago. Active 7 years, 10 months ago. Viewed 3k times. Improve this question. Kara 5, 15 15 gold badges 49 49 silver badges 56 56 bronze badges. Add a comment. Active Oldest Votes. It is very configurable and should work with any VoIP provider. It is based on customized sipXtapi client library and wxWidgets 2. Ekiga uses both the H. It supports many audio and video codecs, and is interoperable with other SIP compliant software and also with Microsoft NetMeeting.
It can run on windows or linux. Figure 3 - Conference call. If you want to use your softphone client as a videophone, you will need a web cam and a good video card. You can, of course, make a one-sided video call when only one of the end-points has a web cam installed. In this case one of the softphones will only be able to use the voice chat function while the other can send video too.
Figure 4 shows you, that if both end-points have a web cam, the communicating partners can see each other's faces while communicating. There is also a possibility to disable the web cam, when you do not want your partner to see your face.
Figure 4 - Video phoning. Video communication also needs a reliable Internet connection. Some time ago with low speed Internet video phoning was discursive or simply impossible, but today the audio and video codecs allow you the possibility of using these technologies with lower Internet connection too.
Sometimes there can be problems with the connection like the video is not continuous, there are jumps in the video stream, the voice and the image are not synchronous, etc. These are the most common problems of a softphone video call, but using the proper codec packs you will not likely meet these any more. Video chats can also be made as conference calls, this gives you the possibility of office meetings or lectures even if the participants are far from each other.
The same method as in the case of voice messaging, you can record and send video messages with your software phone. In this case, you also have to compress your recorded video before sending.
Uncompressed video files are too large to send efficiently. There are standards for video and audio compression. For the voice communication between two end-points, both have to use the same communication protocols and at least one common audio codec. All softphones usually use a wide variety of audio codecs of minimum sets of G.
SIP is an application-layer control signaling protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. A motivating goal for SIP is to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network PSTN.
Its focus is call-setup and signaling. The features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ring back tones or a busy signal - are performed by proxy servers and user agents. Implementation and terminology are different in the SIP world but to the end-user, the behavior is similar.
Voice over IP is a set of technologies, protocols and transmission techniques that are used to establish a stable communication between two connected end-points using the Internet. The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, making packets, and transmission as Internet Protocol IP packets over a packet-switched network. On the receiving side, similar steps usually in the reverse order such as reception of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream.
VoIP is available on many smartphones and internet devices so even the users of portable devices that are not phones can still make calls or send SMS text messages over 3G or Wi-Fi. Figure 5 shows the wide variety of devices that can be used with softphone applications.
Some of these use mobile technologies like smart phones or tablets; others use the standard Internet connections. Figure 5 - VoIP communication devices. Voice and video data are analog but the computers can only cope with digital data, so you have to use some standard methods to digitize the analog data before being able to send or even store it. When you want to build a softphone, you will need to cover this matter too.
Make sure, you choose an SDK that provides support for data digitization, so you wont need to do it from the start. Communication over the Internet means that the two end-points send and receive messages - data packets - over the IP protocol.
The packet sending over the Internet has the critical need of compression, as the packet size is limited and the higher number of the packets increases the possibility of packet loss. This means that some of the sent information will be missed, or the whole message will be impossible to reverse on the receiving side. To ensure the lowest possibility of data loss, there are a lot of data compression techniques that make the sent messages as compact as possible.
One of these standard techniques is packed into the G Alaw codec. G A-law is a logarithmic algorithm for compressing audio and video data. It encodes a 13 bit signed linear PCM sample into logarithmic 8-bit sample. The A-law compression enables more quantization levels at lower signal levels. Now you are familiar with the basic terms of the softphone technologies, so you can start to build your own software VoIP SIP client.
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