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⊕
A2D
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An analog voice signal is received
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The signal is converted to a Pulse Code Modulation (PCM) digital stream
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PCM Processing
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Remove echo
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Remove silence
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Tone Detection
detected signaling tones are routed around the CODEC. Most
CODECs garble signaling tones to the point that they are unrecognizable
by the devices they are intended for.
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Remaining PCM samples are forwarded to the CODEC
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The PCM stream is fed into the CODEC
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Voice frames are created
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Most CODECs also compress the PCM stream
| Codec
| Bandwidth
| IP Bandwidth
(2 Way)
| Packet Delay (ms)
| G.711
| 64kbps
| 160kbps
| 1.0
| G.723
| 6.4 Kbps
| 34 Kbps
| 67.5
| G.729a
| 8 Kbps
| 48 Kbps
| 25.0
| iLBC
| 13.3/15 Kbps
|
| 30/20
| | | | |
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In g.729, each frame samples 10ms (80bytes) of voice streams
and compressed to a 10 bytes frame.
| Sampleing Freq
| 8000 Samples/Sec
| Bits Per Sample
| 8 bits
| Bits Per Second
| 64 Kbps
| G.729 Frame Size (ms)
| 10ms
| G.729 Frame Size (bytes) Before Compression
| 80 bytes
| G.729 Frame Size (bytes) After Compression
| 10 bytes
| iLBC Frame Size (ms)
| 20/30ms
| iLBC Frame Size (bytes) Before Compression
| 160/240 bytes
| iLBC Frame Size (bytes) After Compression
| 30/40 bytes
| | | | | | | | |
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ITU-T G.729語音壓縮標準技術利用 Conjugate-Structure Algebraic
Code Excited Linear Prediction (CS-ACELP)演算法，
所發展出來的技術，其所用語音 frame 為10ms，
所以每一 frame 有80個取樣值(80
bytes)。在每10ms的時間內，語音訊號會被分析，
並利用CELP演算法，取出其特性參數，然後編碼成80 bits 的 frame，
裡面含有代表那段時域訊號對應於CELP的參數，壓縮比為8:1。
G.729語音壓縮標準的應用非常廣泛，如VoIP網路閘道、IP電話、
視訊會議和電話會議等。ITU當初制定G.729語音壓縮標準時，
為了使其具有低位元率、高音質、卻又低複雜度的
特性，在G.729演算法中運用了相當多的專利技術，
其中大部分的專利為國際各大主要電
信業廠商所持有，這些公司包括法國電信(France Telecom)、Universite
de Sherbrooke
及日本電報電話公司(NTT)。它們於1998年3月組織了G.729專利聯盟，
並委由Sipro Lab Telecom公司作為此聯盟的代表，負責處理各專利授權問題。
業者必須事先獲得相關專利
授權許可，方能合法使用生產ITU-T G.729相關產品。
♦
Reference:
U.S. Patent 5664055 ," CS-ACELP speech compression system with
adaptive pitch
prediction filter gain based on a measure of periodicity",
http://www.freepatentsonline.com/5664055.html.
| http://www.itu.int/rec/T-REC-G.729/e
| |
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iLBC (internet Low Bitrate Codec) 是一種免授權的編碼標準。
iLBC在有封包遺失的條件下的其性能明顯優於
G.723、G.728、G.729、GSM等標準編碼。
iLBC是為專為提供穩健的IP語音通訊而開發的語音編碼，
使用8kHz的取樣率。
iLBC編解碼器支持兩種frame size，在13.3kbps位元率下編碼
的 frame size 為30ms，而15.2kbps位元率下編碼
的 frame size 則為20ms。
使用iLBC編碼的封包可以獨立解碼，
IP封包丟時，聲音的損失只侷限在丟失的封包上，
不會影響其他封包的解碼，
因此iLBC抗封包遺失的能力高於其他編碼標準。
♦
Reference:
| http://www.vocal.com/ilbc.html
|
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⊕
Frames to Packets
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Packet Assembler Software within the DSP takes frames from the CODEC
and created packets
several frames may be confined in a single packet
A 12 byte RTP Header is added
The packet is forwarded to the gateway's host processor
The packet is forwarded to the gateway's host processor
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Dialed digits identified by the tone detection performed in the DSP are used
to determin the destination number
This number is mapped to an IP address
| 301-999-1212
| 192.128.100.2
|
A 20 byte IP Header is added to the packet containging:
IP address of this gateway (the source address)
IP address of this destination gateway
An 8 byte UDP header containing source and edestination sockets is also added
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packets are routed in the Intenet
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| Protocol
| Size
(bytes)
| RTP
| 12
| UDP
| 8
| IP
| 20
| Total
| 40
| | | | |
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