1 Improving VoIP Transfer Rate over Internet Syed Misbahuddin, Dr. Engg. Department of Computer Science and Software Engineering University of Hail, Saudi Arabia

2 Presentation outline Research Objective Overview of Voice Over IP (VoIP) Compression algorithm applied to VoIP Problems of compression algorithms The proposed algorithm for Reducing Web delays in VoIP Conclusion

3 Research Objective To investigate an algorithm to Improve VoIP Traffic rate over Internet

4 What is Voice Over IP (VoIP) Using Internet to transfer voice signals after converting them into IP packets

5 Why VoIP voice communication with no or minimal cost through the Internet backbone

6 Digitization of voice signal and 8 bit binary code assignment to each digitized sample Voice signal The sampled voice signals are coded into bits per sec stream The G.729 compression algorithm compresses 64 kb/s to 8 kb/s Compressed voice stream is converted into voice frames of 10 mill sec long carrying 80 bits Generation of Individual Voice Frames in VoIP

7 Transmission of Voice Frame as IP Packet Internet Transmitting end IP Header(20 )UDP Header(8)RTP Header(12)Voice Frame(10) IP Header(20 )UDP Header(8)RTP Header(12)Voice Frame(10) IP Header(20 )UDP Header(8)RTP Header(12)Voice Frame(10) Note: Numbers in each field shows size of field in bytes

8 QoS in VoIP The Quality of Services (QoS) in VoIP technology is related to the short delivery time of the voice data over the Internet To achieve better QoS, data compression algorithms are applied in VoIP systems. Standard voice compression algorithms used are: ITU’s G. 723 and G.729

9 Main Problem with Compression Algorithms The compression algorithms reduce voice quality Better compression algorithms are constantly being investigated to maintain the voice qualities.

10 Justification of Proposing a Data Reduction Algorithm for VoIP The voice signal is analog signal which varies slowly in time. If the voice signal is sampled at relatively high rate, the equivalent digital data will have repeated values in a short time window.

11 Assumptions The voice signals should be sampled at the rate of samples per second each voice sample is assigned a 16 bit code The sampled voice signals are coded into 64 kb/s bit stream A bit stream of 256 kb/s is divided into voice frames of one milli second duration carrying 256 bits Individual voice frames of 1 m sec can be broken into 4 groups of 64 bits of ¼ m sec

12 Sub grouping of voice Frame One Voice Frame of 1 m sec duration of 256 bits 64 bits

13 The Algorithm Store a copy of recently sent voice frame (of 1 m sec) in a buffer called VFT_BUFF Before sending next voice frame, compare contents of subgroups with the content of most recently sent subgroups If in each subgroup some bytes are repeated then produce a 8 bit compression code for each subgroup In compression code the ith bit=1 if ith byte in subgroup is repeated otherwise ith bit=0 if ith byte is new Include compression code as first byte in each subgroup Include non-repeated bytes in each subgroup Send modified voice frame for further processing.

14 Subgroup of voice frame of 64 bits Modified subgroup of voice Frame with compression code (CC) CCNon Repeated bytes 8 bits 0 to 7 bytes Size of modified subgroup=1 byte to 8 bytes

15 Example: Assume 4 initial bytes in each subgroup are repeated and 4 bytes are new Non-Repeated bytes Size of modified Subgroup=5 bytes Compression code

16 Impact of Algorithm on complete voice frame If all bytes in a voice frame are repeated then a 256 bits long voice frame is represented by only 64 bits

17 Voice frame Reconstruction at receiving end of VoIP system The receiving end stores a copy of most recently received voice frame in a buffer called VFR_BUFF When the receiving system receives another voice frame with compression codes in each subgroup then it retrieve the repeated bytes from VFR_BUFF and non-repeated bytes from received voice frame

18 Indication of presence of compression codes in voice frame Two undefined values in Pay Load Type field in RTP header may be used to indicate the presence and absence of compression code in a voice frame

19 Overview of RTP Provides end-to-end delivery services for real-time traffic: interactive audio and video Primarily designed to support multiparty multimedia conferences, typically assumes IP multicast.

20 RTP Header

21 Pay Load Type Field in RTP Header

22 PT Values in RTP Header for indicating the presence and absence of compression code in Voice Frame PT =16 Normal Voice Frame PT=17 Voice Frame containing compression codes

23 Summary of Data Reduction Algorithm Begin Very first Voice frame Transmitting End Yes Save copy in VF_TBUF Add RTP, UDP and IP Send VF to Internet Obtain VF No Compare VF with VF_TBUF Repetition in VF No Update VF_TBUF Append compression code in each subgroup in VF. Modify PT in RTP Yes Update VF_TBUF

24 Summary of Data Reduction Algorithm Receiving End Begin Obtain VF Very first Voice frame Yes Save copy in VF_RBUF Process VF No CC in VF Yes No Save copy in VF_RBUF Retrieve repeated bytes in subgroup from VF_RBUF Retrieve non-repeated bytes form received VF

25 Conclusion With the application of proposed data reduction algorithm, off the shelf data compression algorithms may not be needed Proposed data reduction algorithm may give better Internet bandwidth utilization retaining the quality of voice signals