project presentation “ analyzing factors that affect voip call quality ” presented by: vamsi...
TRANSCRIPT
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PROJECT PRESENTATION“Analyzing Factors that affect VoIP Call
Quality”
Presented By:Vamsi
Krishna Karnati11/24/2014
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CONTENTS
• Factors affecting VoIP call quality
• Overview of Audio Codecs used in this project.
• Codecs comparison.
• Bandwidths calculation.
• Results
• Comparing results with theoretical values.
• Problems Faced.
• Conclusion
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Factors affecting VoIP call Quality
• Audio Codec• All VoIP telephony systems uses codecs to compress and
decompress audio signals on either ends. Ex:G.711,G.722,G.726,G.729,GSM,iLBC.
• In voice call, Higher compression lead to less data transfer, but high compression causes quality degradation.
• Compression rates like 8Kbps, 13Kbps , 64Kbps, etc; are only for audio and protocol overheads are added over to it.
• Also with complicated implementation of codec leads to more usage of CPU resources.
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Factors effecting VoIP call Quality(Cont.)
• Latency:• Depends on distance packet travels and network conditions.
• Delay increases with increase in router hops.
• Delay of more than 150ms causes the caller notice delay.
• Jitter:• Packets start with equal spacing , but at receiving end packets receive at
different spacing's.
• Also may not arrive in same order as sent
• Jitter buffer acts then, to rearrange them for decompression.
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Factors effecting VoIP call Quality(Cont.)
• Packet Size:• Higher packet size, overall BW is reduced
• So as BW reduces, sending bigger packets is preferred.
• Packet Loss:• Collisions cause packet loss most of the times.
• Codecs perform actions to compensate lost packets
• More than 5% packet loss, lower voice quality.
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Overview on Codecs
• G.711:• PCM at rate 8000samples/sec & 8 bits per sample. (64kbps)
• No compression implemented.
• G.722• Codec based on ADPCM
• Improved quality of speech, due to 50-7000Hz speech BW
• Works well in LAN’s, where BW is high http://www.itu.int/rec/T-REC-G.722/e
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Overview on Codecs(Cont.)
• G.726:• Transmits at different rates like 16,24,32,40 Kbit/s
• Commonly used is 32 kbps.
• Network capacity is doubled when compared with G.711 with rate 64Kbps.
• GSM:• Codec designed by European TSI, GSM mobile networks.
• Compress frame to 33 Bytes, as it operate on 20ms frames.[13 kbit/s]
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Overview on Codecs(Cont.)
• iLBC:• Free codec for robust call quality.
• With frame length of 30ms, it results bit rate of 13.33kbps
• G.729:• Requires less bandwidth and provide high quality audio[MOS=4]
• Each 10ms frame is encoded to 10 bytes [Bit rate =8kbit/s]
• It is licenced
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CODECS USED
CODECS BIT-RATE
G.711 64kbps
G.722 64kbps
G.726 32kbps
iLBC 13kbps
G.729 8kbps
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Bandwidth Calculations
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I/O Bandwidth Graphs
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• From the results:
• BW: Bandwidth = (packets/second x [packet size] bytes/packet x 8 bits/byte)
• G.711 =162.048kb/sec 1.21MB/Min
• G.722= 171.296kb/sec 1.28MB/Min
• G.726 =108 kb/s 0.81MB/Min
• This is the Total BW for full duplex call(BW of incoming and outgoing)
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Codec Practical BW Theoretical BW
G.711 162.048Kbps 174.4Kbps
G.722 171.29Kbps 159.26Kbps
G.726 – 32kbps 108.Kbps 110.4Kbps
iLBC 164Kbps
G.729 {Licensed} 62.4Kbps
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Results
• Based on BW calculations, I found G.726 is efficient codec.
• G.729 is more efficient to it according to theoretical values.
• I understood, Compression will help more in WAN reducing BW, rather than in LAN where high BW is available.
• In case of LAN high BW codec provides better
Voice quality. [g.711]
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• Jitter:• This was other QOS factor compared for VoIP call made.
• Results show no much difference in mean jitter values
• Where G.722 having less when compared to other codecs performed. [0.82ms]
• Only when jitter exceeds 150ms, the user notice the delay.
• Packet Loss:• No packet loss observed among 3 codecs, as they are on same network
• This is also a major factor in terms of WAN, so less BW codec is preferred
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PROBLEMS FACED
• Tried to download G.729 for asterisk.• Not possible as it is licensed .
• Used softphones to make call using GSM codec.• Call was successful, but unable to capture RTP packets.
• I could finally resolve it using the IP address of Computer
for calling
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CONCLUSION
• I would conclude saying that:• Using G.711 in the network with low BW is a bad choice.
• It will suffer from BW limitation and Packet loss.
• So, using Codec like G.729 for WAN and G.711 for LAN is preferred.
• At last, the choice of codec will improve the call conversation quality.
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REFERENCES
• http://www.itu.int/rec/T-REC-G.722/e
• http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-UnderstandingVoIP-SECT-3.html
• http://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html#topic1
• http://www.techtarget.com/
• http://whirlpool.net.au/wiki/voip_codecs
• http://www.voip-sip.org/typical-voip-problems/
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