What is G.711 codec

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The G.711 codec algorithm defines rules for encoding/decoding non-compressed waves. The input data is derived from PCM data whose sampling frequency and bit rate are 8000 Hz and 64 kbit/s respectively. Based on the companding rate adopted for signal quantizing, the G.711 algorithm is classified into G.711 A and G.711 μ. The difference between the two lies in the fact that G.711 A uses a companding curve of 13-fold lines, whereas G.711 μ uses a companding curve of 15-fold lines.
The G.711 coder and decoder use a simple algorithm and compress signals by quantizing them in a nonlinear method, providing a short delay and high voice quality. However, the G.711 coder and decoder have a high data rate, which makes them sensitive to errors in transmission channels.
UMG8900 supports the G.711 voice codec mode and the switchover between G.711 A and G.711 μ, and can interwork with PSTN networks.

Other related questions:
What is G.711
G.711, also known as Pulse Code Modulation (PCM), is a commonly used waveform codec. G.711 uses a sampling rate of 8,000 samples per second. Non-uniform quantization (logarithmic) with 8 bits is used to represent each sample, resulting in a 64 kbit/s bit rate. G.711 defines two main compression algorithms, the ?-law algorithm (used in North America & Japan) and A-law algorithm (used in Europe and the rest of the world). A-law encoding takes a 13-bit signed linear audio sample as input. μ-law encoding takes a 15-bit signed linear audio sample as input. G.711 is simple, and provides short delay and high audio quality. G.711 has high data rate and is sensitive to errors on channels. The UMG8900 supports G.711, A/μ conversion, and interworking with the PSTN.

What are the differences in the implementation mode between AAC-LD stereo and G.722 mono codecs on an endpoint?
G.722 is a monaural protocol. Currently, the ViewPoint 9000 series endpoint supports only the following stereo protocols: AAC-LD and HWA-LD. If the negotiation result is mono, the input audio channel with the higher volume is selected from the left and right input audio channels, the sound collected by this audio channel is transmitted to the peer site, and both the left and right audio channels at the remote site play this sound. If the negotiation result is stereo, both the left and right input audio channels are enabled on the web interface of the endpoint, and both the left and right input audio channels have sound input, sound is transmitted to the peer site, and both the left and right audio channels at the peer site play sound. If the negotiation result is stereo, both the left and right input audio channels are enabled on the web interface of the endpoint, but only the left input audio channel has sound input, sound is transmitted to the peer site, the right audio channel at the peer site plays sound, but the left audio channel at the peer site plays no sound. If the negotiation result is stereo, only the left input audio channel is enabled on the web interface of the endpoint, and only the left input audio channel has sound input, sound is transmitted to the peer site, and both the left and right audio channels at the peer site play sound.

Is it possible that the DTMF secondary dial tone cannot be detected if the G.729 codec mode is used
The DTMF second dial tone may not be detected because the G.729 codec mode is a lossy codec mode.

Codecs supported by IP phones
Codec is used for interconnection between wideband and narrowband. On the U1900 series unified gateway, run the show system information command to check the supported codec modes (whose priorities are in descending order). Assume that the network is as follows: IP phone -- U1900 -- (Narrowband trunk) -- PSTN The U1900 supports and has the following codec modes enabled: G711A | G711U | G729 | G723 | AMR_WB | iLBC_1520 | iLBC_1333 | G722 | G7221_24 | G7221_32. The IP phone supports and has the following codec modes enabled: G711U | G711A | G729 | G722. - When the IP phone makes an outgoing call through the narrowband trunk, the U1900 compares its codec modes with those of the IP phone after receiving the call request. The U1900 detects that the G711U | G711A | G729 modes of the IP phone are the common codec modes, and selects G711U with the highest priority to set up the call. (The priority relationship of the IP phone is used, which is called peer preference. The U1900 supports only the peer preference mode.) - When the narrowband trunk side makes an outgoing call to the IP phone, the process is similar. The IP phone compares its codec modes with those of the U1900 after receiving the call request. The IP phone detects that the G711A | G711U | G729 modes of the U1900 are the common codec modes, and selects G711A with the highest priority to set up the call. The IP phone also supports only the peer preference mode.

How to modify the codecs and their priorities for the U1900 series unified gateway
1. You can modify the codecs of the U1900, but their priorities cannot be set using the CLI. The codec to use during a call is determined through negotiation. 2. You can run the config system codetype G711A&G711U&G729&G723&iLBC_1520&iLBC_1333 command in config mode to set the voice codecs.

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