PROJECT PRESENTATION “ Analyzing Factors that affect VoIP Call Quality ” Presented By: Vamsi Krishna Karnati 11/24/2014

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 Hz speech BW Works well in LAN’s, where BW is high

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 kbps G kbps G kbps 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 = kb/sec 1.21MB/Min G.722= kb/sec1.28MB/Min G.726 =108 kb/s0.81MB/Min This is the Total BW for full duplex call(BW of incoming and outgoing)

CodecPractical BWTheoretical BW G Kbps174.4Kbps G Kbps159.26Kbps G.726 – 32kbps108.Kbps110.4Kbps iLBC164Kbps 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 UnderstandingVoIP-SECT-3.html UnderstandingVoIP-SECT-3.html consume.html#topic1 consume.html#topic