G.7?? (ulaw, alaw)

由 datahunter 在二, 09/10/2012 - 00:27 發表

最後更新: 2015-07-22

查看支援什麼 codec

core show codecs

2 audio     ulaw (G.711 u-law)
3 audio     alaw (G.711 a-law)
36 video      vp8 (VP8 video)
..................

G.711(OpenSource)

* released for usage in 1972
* PCM
* fax communication over IP networks
* 64 kbit/s
* 300–3400 Hz
* 8,000 samples / 8 bits each sample
* Typical algorithmic delay is 0.125 ms
* logarithmic
* almost zero load on the CPU

http://en.wikipedia.org/wiki/Coherence_bandwidth
http://en.wikipedia.org/wiki/Narrowband

Both are logarithmic:

μ-law algorithm (used in North America & Japan) Input: 13-bit ---> 8-bit
a-law algorithm (used in Europe and the rest of the world) Input: 14-bit ---> 8-bit
(specifically designed to be simpler for a computer to process)

G.726

ADPCM (Adaptive Differential Pulse-Code Modulation)
16 Kbps, 24 Kbps, and 32 Kbps
it sends only enough information to describe the difference between the current sample and the previous one
inability to carry modem and fax signals

# 經測試, 質量很差 @@"

* Asterisk(1.8) currently supports the 32kbps standard only.

G.729 (Proprietry)

# DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm.

運作
compresses digital voice in packets of 10 milliseconds duration

特性:
bit rate of 8 kbit/s
extended: G.729a and G.729b

P.S.
Dual-tone multi-frequency signaling (DTMF), fax transmissions, and high-quality audio cannot be transported reliably with this codec.

# put in your Asterisk modules directory.
codec_g729.so
codec_g723.so

GSM (Proprietry)

audio into between 6.5( Half Rate ) and 13 kbit/s( Full Rate)

GSM (Global System for Mobile communications)
* GSM includes a codec
* linear predictive coding

運作:

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.

Newer:

HR (Half Rate) uses CELP-VSELP (Code Excited Linear Prediction - Vector Sum Excited Linear Prediction)

Enhanced Full Rate (EFR) uses ACELP (Algebraic Code Excited Linear Prediction)(12.2 kbit/s) (yr 1997)

後繼: variable-rate codec: AMR-Narrowband

asterisk 1.8

gsm = 13 Kbps (full rate), 20ms frame size

speex (Open Source)

* bitrates ranging from 2 to 44 kbps
* based on CELP
* Narrowband (8 kHz), wideband (16 kHz), and ultra-wideband (32 kHz) compression in the same bitstream
* Packet loss concealment
* Variable bitrate operation (VBR)
* Voice Activity Detection (VAD)
* Discontinuous Transmission (DTX)
* Fixed-point port
* Acoustic echo canceller
* Noise suppression

PESQ MOS-LQO (Perceptual Evaluation of Speech Quality)
a test methodology for automated assessment of the speech quality as experienced by a user of a telephony system.

http://en.wikipedia.org/wiki/PESQ

# 系統可支援的編碼類型
CLI> core show codecs [audio|video|image]

# 系統可用的語音編碼及轉換所需要的時間
CLI> core show translation

CLI> core show translation recalc 10
重新計算不同語音編碼轉換所需的時間

Check SIP 通話所使用的語音編碼

server*CLI> sip show channels

Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry     Peer
192.168.88.172   (None)           7100491849bf009  0x0 (nothing)    No       Rx: REGISTER               <guest>
192.168.88.171   201              5e0634bc697052d  0x2 (gsm)        No       Rx: ACK                    201
2 active SIP dialogs

server*CLI> sip show channel 5e0634bc697052d

  * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                [email protected]
  Owner channel ID:       SIP/201-00000007
  Our Codec Capability:   0x2 (gsm)
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   0x2 (gsm)
  Joint Codec Capability:   0x2 (gsm)
  Format:                 0x2 (gsm)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.88.171:5060
  Received Address:       192.168.88.171:5060
  SIP Transfer mode:      open
  Force rport:            No
  Audio IP:               192.168.88.150 (local)
  Our Tag:                as15c335d0
  Their Tag:              16035868
  SIP User agent:         CM5K-PHONE  (903060)
  Username:               201
  Peername:               201
  Original uri:           sip:[email protected]:5060
  Caller-ID:              201
  Need Destroy:           No
  Last Message:           Rx: ACK
  Promiscuous Redir:      No
  Route:                  sip:[email protected]:5060
  DTMF Mode:              rfc2833
  SIP Options:            100rel replaces replace
  Session-Timer:          Inactive

MaxCallBR

Default is 384 kbps

P.S.

Your call for that peer allowed max bit rate or bandwidth of 384 kpbs only

sip.conf

;maxcallbitrate=384

設定語音編碼