Performance Analysis of Different Codecs in VoIP Using SIP

Performance Analysis of Different Codecs in VoIP Using SIP

Ravi Shankar Ramakrishnan and P. Vinod kumar
1 2

International Institute of Information Technology E- mail: 1 itsravishankar84@gmail.com, 2pvknet@gmail.com, 2vinodk_june07@net.isquareit.ac.in

ABSTRACT: Converged IP networks seek to incorporate voice, data, and video on the same infrastructure. However, the integration of all types of traffic onto a single IP network has several advantages as well as disadvantages. While reducing cost and increasing mobility and functionality, VoIP may lead to reliability concerns, degraded voice quality, incompatibility, and end- user complaints due to changing network characteristics. The main purpose of V oIP, Various CODECS used in V oIP and packet loss, Jitter, delay are analyzed and discussed. Keywords Converged IP Networks, Reliability Concerns, VoIP, CODECS.

INTRODUCTION

INTRODUCTION TO SIP
Session Initiation Protocol (SIP) is a peer-peer signaling protocol for VoIP, developed by the IETF MMUSIC Working Group and defined in RFC 2543. It is a proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality. SIP requires a simple core network with intelligence embedded in endpo ints; thus it is highly scalable. It closely resembles HTTP and SMTP; thus SIP sits comfortably alongside Internet applications.

oice over IP (VoIP) is the ability to transmit speech over packet-switched IP networks. VoIP is an acronym for Voice over Internet Protocol, or in more common terms phone service over the Internet. If you have a reasonable quality Internet connection you can get phone service delivered through your Internet connection instead of from your local phone company. Placing a phone call using VoIP will c reate a digital signal from the analog input, place those digital signals into packets with source and destination network addresses and finally send the information over the Internet or internal company IP networks, thus bypassing the need for the PSTN lines. However, it is important to note that both source and destination terminals must support the particular codec for proper encoding and decoding. Two popular VoIP standards are H.323 and Session Initiation Protocol (SIP). Both standards allow for direct call establishment between VoIP-capable terminals or the use of gatekeepers, which can be used to negotiate connections between endpoints. They can be used to translate between IP addresses and telephone numbers, perform registration and authentication functions, and manage bandwidth.

SIP-Allied Protocols
SIP interoperates with: SDPTo describe the payload of message content and characteristics SAPfor advertising multimedia session via multicast RSVPTo reserve network resources for providing QoS RTPFor real-time transmission RTSPfor controlling delivery of streaming media RADIUS For authentication LDAPFor location discovery.

SIP Call Flow

Fig. 1: Basic VoIP Implementation

Fig. 2: SIP Call Flow Diagram

Performance Analysis of Different Codecs in VoIP Using SIP

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SIP Protocol Architecture

Codec Comparison
The following table lists the various codecs used in voice over IP, and in particular SIP.
Codec G.711 G.722 Sampling Rate (KHz) 8 16 16 16 8 8 8 8 8 8 unknown 8 Bandwidth (Kbps) 64 48 56 64 5.3 6.3 16 24 32 40 16 8 License Open Source Open Source

G.723.1 G.726

Proprietary Open Source

G.728 G.729 Fig. 3: SIP Protocol Architecture

Open Source Patented

QUALITY OF SERVICE (QOS)

In networking, quality can mean many things. In VoIP, quality simply means being able to listen and speak in a clear and continuous voice, without unwanted noise. QoS (Quality of Service) is a major issue in Vo IP implementations. The issue is how to guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due interfer ence from other lower priority traffic. Things to consider are Latency, Jitter and Packet loss.

REAL TIME PROTOCOL (RTP)

Audio, video, and multimedia services require the use of RTP, which provides the necessary end-to-end delivery requirements of time sensitive data. Both RTP and RTCP were designed to run independently of the underlying transport and network layers. RTP often runs in unison with the User Datagram Protocol (UDP).

RTP packet format

A typical RTP packet includes a sequence number that allows the receiver to reconstruct the data the sender has sent in the appropriate order.

MEAN OPINION SCORE (MOS)

Quality Scale Excellent Good Fair Poor Bad Score 5 4 3 2 1 Listening effort Scale No effort required No appreciable effort required Moderate effort required Considerable effort required No meaning understood with reasonable effort

CODEC
A codec, which stands for coder-decoder, converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay. It's the essence of Vo IP. It converts each tiny sample into digitized data and compresses it for transmission.

Coding techniques are such that speech quality degrades as data rate reduces. However, the relationship is not linear.
Codec G 711 G 726 G 726 G 728 G 729 GSM Data Rate 64 32 63 16 8 13 Mos Score 4.3 4.0 3.8 3.9 4.0 3.7

Common VoIP Codec

Codec G.711 G.722 G.723.1 G.726 G.729 Comments Delivers precise speech transmission. Needs at least 128 kbps for two -way. Adapts to varying compressions and bandwidth is conserved with network congestion. High compression with high quality audio. Lot of processor power. An improved version of G.721 and G.723 (different from G.723.1) Excellent bandwidth utilization. Error tolerant. License required.

RESULTS
The comparison between three different codecs has been analyzed by implementing peer-t o-peer VoIP network using SIP server and caller and callee.

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Mobile and Pervasive Computing (CoMPC2008)

SIP Server Active Calls

CALLEE: Delay Parameter

Fig. 4: SIP UAS Active Call timing Diagram

Fig. 7: Voice application traffic delay

CALLEE: Jitter Parameter

Fig. 5: SIP UAS Active Call Bar Diagram

Fig. 8: Voice traffic delay variation

CALLEE: Voice Traffic Received

CALLER: Voice Traffic Received

Fig. 6: Voice application traffic received.

Fig. 9: Voice application traffic received

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CALLER: Delay Parameter

CONCLUSION
Thus we have described the various codecs in VoIP implementation and analyzed three commonly used codecs using peer-to-peer network scenario. These are common narrow band codecs. It can be analyzed from the results that G.711 is an ideal solution for PSTN networks with PCM scheme. G.723 is used for voice and video conferencing however provides lower voice quality. Music or tones such as DTMF cannot be transmitted reliably with G.723 codec. G.729 is mostly used in VoIP applications for its low bandwidth requirement.

Fig. 10: Voice application traffic delay

REFERENCES
[1] Quality of Service for voice over IP- Cisco documentation. [2] The effect of dynamic voice codec selection for active calls on voice qualityThesis by Jered Daniel Ast. [3] Packet Scan Users GuideGL Communication Inc. [4] VoIPLecture notes by David Wang. [5] Measurement Challenges for VoIP infrastructuresPrasad Calyam, Internet2 VoIP Workshop. [6] VoIP as a collaborative tool on client PCIntel White paper.

CALLER: Jitter Parameter

Fig. 11: Voice traffic delay variation