To use the TalkLah system, there are a few requirements to be met:
- High speed, stable Internet connection. The minimum requirement is 64 Kb/s in each direction or a total of 128kb/s. To verify connection reliability you can use packet tracing software to reveal network problems. We recommend a product called "Ping Plotter". You can download a free trial version at http://www.pingplotter.com.
- SIP IP phone, you can use a softphone if your PC has speakers and microphone or a headset.
- A personal computer for account management.
- SIP IP phone must use G729 as the low bandwidth CODEC since TalkLah does not support G723.
G.711
G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972.
G.711 is a standard to represent 8 bit compressed pulse code modulation (PCM) samples for signals of voice frequencies, sampled at the rate of 8000 samples/second. G.711 encoder will create a 64 kbit/s bitstream.
There are two main algorithms defined in the standard, mu-law algorithm (used in North America & Japan) and a-law algorithm (used in Europe and the rest of the world). Both are logarithmic, but the later a-law was specifically designed to be simpler for a computer to process. The standard also defines a sequence of repeating code values which defines the power level of 0 dB.
The equations are:
mu-law:
y = ln(1 + ux) / ln(1 + u) with u = 255
A-law:
y = Ax / (1 + ln A) for x <= 1/A where A = 87.6
y = (1 + ln Ax) / (1 + ln A) for 1/A <= x <= 1
G.729
G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. Music or tones such as DTMF or fax tones cannot be transported reliably with this codec, and thus use G.711 or out-of-band methods to transport these signals.
G.729 is mostly used in Voice over IP (VoIP) applications for its low bandwidth requirement. Standard G.729 operates at 8 kbit/s, but there are extensions, which provide also 6.4 kbit/s and 11.8 kbit/s rates for marginally worse and better speech quality respectively. Also very common is G.729a which is compatible with G.729, but requires less computation. This lower complexity is not free since speech quality is marginally worsened.
The annex B of G.729 is a silence compression scheme, which has a VAD module which is used to detect voice activity, speech or non speech. It also includes a DTX module which decides on updating the background noise parameters for non speech (noisy frames). These frames which are transmitted to update the background noise parameters are called SID frames. A comfort noise generator (CNG) is also there, because in a communication channel, if transmission is stopped, because it's not speech, then the other side may assume that link has been cut. This is also taken care of by the annex B standard.
G.726
G.726 is ITU-T speech codec operating at bit rates of 16-40 kbit/s. The most commonly used mode is 32 kbit/s, since this is half the rate of G.711, thus increasing the usable network capacity by 100%. G.726 is based on ADPCM technology. ITU standardized G.726 for the first time in 1984. Several additions to the standard have been done later. The additions include additional modes (originally G.726 was only 32 kbit/s), elimination of all-zero codewords.
G.723
G.723 is a ITU-T standard wideband speech codec. This is an extension of Recommendation G.721 adaptive differential pulse code modulation to 24 and 40 kbit/s for digital circuit multiplication equipment application. Superceded by G.726, this standard is obsolete.
ILBC
The Internet Low Bit Rate Codec (iLBC) is a royalty free narrowband speech codec, developed by Global IP Sound (GIPS). It is suitable for VoIP applications, streaming audio, archival and messaging. The algorithm is a version of block-independent linear predictive coding, with the choice of data frame lengths of 20 and 30 milliseconds. The encoded blocks have to be encapsulated in a suitable protocol for transport, eg. RTP.
GSM
GSM (Global System for Mobile communications) is a cellular phone system standard popular outside the USA.
GSM includes a codec, often just referred to as the GSM when discussing codecs.
The original 'Full Rate' GSM speech codec is named RPE-LTP (Regular Pulse Excitation Long-Term Prediction). This codec uses the information from previous samples (this information does not change very quickly) in order to predict the current sample. The speech signal is divided into blocks of 20 ms. These blocks are then passed to the speech codec, which has a rate of 13 kbps, in order to obtain blocks of 260 bits.
