PROJECT PRESENTATION“Analyzing Factors that affect VoIP Call

Quality”

Presented By:Vamsi

Krishna Karnati11/24/2014

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

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.

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.

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.

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

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]

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

CODECS USED

CODECS BIT-RATE

G.711 64kbps

G.722 64kbps

G.726 32kbps

iLBC 13kbps

G.729 8kbps

Bandwidth Calculations

I/O Bandwidth Graphs

• 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)

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

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]

• 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

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

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.

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/