Hi,
With FP11 due out in October and G.711 audio codec support added to this, will streaming G.711 through Wowza then work ‘automatically’? Or must support for this then still be developed?
Peter,
Yes, if the client (Flash 11) supports it. Wowza reflects streams. Wowza 3 transcoder will be able to transcode g.711 audio to AAC for clients (iOS, Android, Flash <11) that do not
Richard
Peter,
Yes, if the client (Flash 11) supports it. Wowza reflects streams. Wowza 3 transcoder will be able to transcode g.711 audio to AAC for clients (iOS, Android, Flash <11) that do not
Richard
To optimize the telephony the G711 aLaw codes is perfect. You can modify the dynamic range of an analog signal for digitizing as well. This is a logarithmic algorithm and it has been designed to be simpler for computer processing than uLaw algorithm. It also provides a more dynamic range resulted in a better sound quality because sampling artifacts are better suppressed. It has a very low processor requirements and needs at least 128 kbps for two-way.
According to my opinion, in communication it is vital to use the best possible solutions that provide the quality and excellence. For this reason, the solution of G711 codec, also known as Pulse Code Modulation (PCM) is a frequently used waveform codec. The G711 uses a sampling rate of 8000 samples/sec, with a tolerance of 50 parts per million (ppm). The G711 codec comes in two different compression algorithms: µ-law and A-law.
The G711 A-law compression algorithm is used in Europe, and almost all over the world. The A-law is logarithmic, and lighter for the computer to process. The G711 A-law encodes a 13 bit signed linear PCM sample into logarithmic 8-bit sample. As a result, the G711 encoder will be able to produce a 64 kbit/s bitstream for a signal that is sampled at 8 KHz. The A-law compression enables more quantization levels at lower signal levels. A 13-bit signed linear audio sample as input is converted to an 8 bit value as follows:
G.711 is an ITU-T standard algorithm for audio companding that is used for digital communication systems and supported by most of VoIP providers. G.711 codec provides the best voice quality for VoIP. Since it uses no compression it sounds like a regular or ISDN phone and it ensures a lowest latency. However it takes more bandwidth than other codecs. It defines two slightly different algorithms: A-Law and U-Law.
I wish to share a useful article about the topic. If you are interested you find it here:
http://voip-sip-sdk.com/p_214-g711-alaw-codec-voip.html
Best Wishes,
NikoJarvi
