Bandwidth Comparison for SIP and IAX Protocol through Asterisk

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1 Bandwidth Comparison for SIP and IAX Protocol through Asterisk Prof. Priyanka Kadam SIES-GST Nerul, Navimumbai Dr. Zia Saquib ABSTRACT Initially Internet is used for data communicationbut now a days with the help of VoIP technology we are able to transmit voice and video streams over the internet. This paper describes IP-PBX server based on Asterisk. Asterisk is one of the first open source PBX software. Traditional EPABX system have many disadvantages over Internet Protocol Private Branch Exchange (IP-PBX). This paper contain brief introduction about SIP and IAX protocol along with its channel configuration in Asterisk. Bandwidth comparisons of SIP and IAX calls with two different codecs G.711 and GSM is done by online erlag s B software. Keywords VoIP, SIP, IAX, EPAX, IP-PBX, RTP, UDP INTRODUCTION For telephony communication most of the organizations uses Electronic Private Automatic Branch Exchange (EPABX). This telephony communication is used by internal employees and with outside world. PBX is a device which shares phone lines, and user can connect extension to it. It works as mini telephone exchange. In such a system extensions are connected to the copper cables to the central electronic system. Two telephone lines in EPABX can be divided in to eight lines which are used for incoming and outgoing lines [1]. But the disadvantage of EPABX system is it requires lots of maintenance work and extra wiring is needed as we add new extension, and it is less flexible and less secure in nature. To Overcome the drawbacks of traditional EPABX system is replaced by IP-PBX[2]. During last decades Voice over Internet Protocol (VoIP) has become a very widespread technology [3]. Due to the advancement of VoIP protocols and codecs it gives drastic change in media transmission. All there advancements are possible because of evolution of circuit switch technology to efficient packet switch technology [4]. By using computers, laptops, IP phones and smart phones with internet facility we can implement IP- PBX system. The advantage of IP-PBX is it reduces time of installation and configuration. It is easier to install and configure than proprietary phone system. Biggest economic advantage of IP-PBX is cost efficient system and it replaces traditional telephone communication system by single line to each user. Hence it is more flexible and reduced maintenance cost. This paper contain brief introduction about SIP and IAX protocols along with its channel configuration in Asterisks. Main focus of this paper is to compare bandwidth requirement of SIP and IAX protocol for multiple calling. This comparison is done by selecting two most popular codecs i.e. G.711 and GSM. VOICE OVER INTERNET PROTOCOL (VoIP) Initially IP network is known for data communication but as technology grows now a day we are using same IP network for voice and video communication. This voice communication through internet is known as VoIP. As VoIP network based on packet switched technology hence its bandwidth requirement is minimum because packets are sent only when necessary. But in case of PSTN network it demands constant 47 Prof. Priyanka Kadam, Dr. Zia Saquib

2 availability of communication channel. Other advantages of packet switched network are it is easier to compress the data at the same time it increases the efficiency of medium. Packet switched network has its own disadvantages more specifically in case of latency and buffering. To obtain maximum bandwidth utilization before transmission in the network voice packets are generally buffered together. Internet protocol is known for best effort delivery transmission, i.e. there is no guarantee about the quality of service, which means it does not sure that delivered in correct order, or whether they are arriving with in a required time bound or not. Many time voice packets may arrive at variable rate, this variable time delay is known as jitter and this is needed to be fixed at receiver side. To fix this jitter most commonly adapted method is buffering of received data before playing. But this buffering may leads to the problem of increasing the latency, which ultimately decreases the quality of communication. PROTOCOLS USED IN VoIP A] Session Initiation Protocol:- Sip is leading protocol for VoIP communication. It has been developed by IETF hence it is unique because all other VoIP protocols are developed by ITU. SIP generally set up, modify and tear down the multimedia session between two clients. In case of VoIP communication through SIP, media stream is separated by the signaling messages. Mobility is one of the most important advantages of SIP. In SIP communication stream is associated with the user only and does not have any relation with device being used. Hence SIP user can register his location from any device with internet connection and then media stream is routed to the particular device on which user is registered. In SIP RTP is used to carry media, but it does not define any specific transport. SIP separately uses Session Description Protocol which gives detail about media content for example codec, IP address and port etc[7]. As SIP is known as client Server protocol, but its terminating points contain User agent Client (UAC) and User Agent Server (UAS). SIP requests are generated by User Agent Client and Use r Agent Server gives replies on the request these user agent servers can be SIP proxy server. Remapping of SIP addresses to other addresses is done by SIP Redirect Server, which return s these addresses to the clients. B] Inter Asterisk Protocol:- Initially IAX is a proprietary protocol, Mostly it is deployed in Digium equipment s. But now a days its second version which is developed to support Open Source Asterisk PBX. Originally IAX protocol is developed for communication between two asterisk servers. As IAX user single UDP stream for communication hence it is consider as its biggest advantage. This UDP stream contains signaling as well as multimedia information. Unlike from other protocols IAX does not use any RTC protocol which provides encapsulation to carry the media. Hence this feature of separate UDP stream makes administrator operation more easier like firewalls and network Address translations (NAT). Due to these features IAX administrator have to face less problems as compare to other streaming protocols. Truncking is another advantage of IAX protocol. This truncking can be done by encapsulating the data for more than one call in a single packet. This feature of IAX reduces the IP overhead which altimetry reduces the latency, hence it also require lower bandwidth without compromising Audio quality of a call. ASTERISK Asterisk is one of the most popular Open Source Software PBX. A PC or headwear device with properly installed Asterisk server along with its correct interfaces is able to provide all the functions of full featured PBX. Asterisk is open source and also propriety protocol. User is free to use, modify and change asterisk suit according to the requirements. Asterisk gives some advance features like: Music on hold, it supports media streams and mp3 files, monitors the call queues. It is integrated service with text-to-speech and voice recognition, and also provides PSTN connectivity through Aserisk. 48 Prof. Priyanka Kadam, Dr. Zia Saquib

3 Asterisk supports many VoIP protocols like SIP, H.323, IAX, MGCP, SCCP (Cisco skinny) and Nort el unistim [8]. Out of these protocols, this paper mainly focuses on the comparison between SIP and IAX protocols. With the help of asterisk VoIP calling we are trying to calculate bandwidth requirement. And the parameters through witch bandwidth calculation varies are specified as follow, A] Asterisk Supported Channels:- Chan_sip:- It supports VoIP communication by using SIP protocol. Is Dial string is sip/channel. Chan_iax:- It supports VoIP communication by using IAX protocol. Its dial string is iax/channel. B] Asterisk Supported Codecs:- Codec is defined as a device which encodes as well as decodes the speech signal at transmitter and receiver side. Mainly codecs are used for speech coding purpose and also they give speech compression. Following are some examples of codecs which are supported by Asterisk. WORKING OF SIP AND IAX WITH ASTERISK In case of VoIP communication through SIP Asterisk work as a register and location server i.e. it connects minimum two User Agent Clients (UACs) to itself. Because of this asterisk is considered as a Backto-Back user agent (B2BUA). In other words asterisk act as a bridge in between two SIP channels. Once the connection is establish in between the sip channel they can communicate together by using SIP proxy like Sip Express Router (SER). This mechanism is possible because sip protocol supports re-invite function. A. SIP IP PHONE CONFIGURATION:- SIP channel is configured in the /etc/asterisk/sip.conf directory. This sip.conf directory contains all the parameters related to the SIP phone and VoIP providers. Before calling SIP client have to be configured. 1. GENERAL SECTION IN SIP.CONF Allow/disallow: It defines codecs selection. Bindaddr: This is a particular IP address which is bonded to the asterisk SIP listener. Default bind address is If default bind address is set then it can be connected to all other interfaces. Fig:1 SIP.conf Channel configuration. 49 Prof. Priyanka Kadam, Dr. Zia Saquib

International Journal of Innovations & Advancement in Computer Science context: Unless there is change in client section, sets the default context for all clients. bindport: SIP UDP port to listen.

4 International Journal of Innovations & Advancement in Computer Science context: Unless there is change in client section, sets the default context for all clients. bindport: SIP UDP port to listen. maxexpirey: It is maximum time to register (seconds). Other parameters are optional in this section. CLIENT SECTION IN SIP.CONF Type: There are three choices available for selecting configuration class Ex. Peer, user and friend. Peer: Asterisk sends calls to a peer. i.e only outgoing calls are allowed User: Asterisk receives calls from a user. i.e only incoming calls are allowed. Friend: Both incoming and outgoing at same time. Host: It is IP address or host name. In general default setting is dynamic, used when the host registers to Asterisk. secret: Password to authenticate peers and users B. IAX IP PHONE CONFIGURATION bindport : Port number to bind or listen. Default bindport for IAX is bindaddr This is a particular IP address which is bonded to the asterisk IAX listener. example: bindaddr= iaxcompact -yes default condition. Iaxcompact- no If we are using layered switches. nochecksumyes - no: Disable UDP checksums. Delayrejectyes- no: To avoide brute force attack and increase security this option should enable. Fig :2iax.conf channel configuration 50 Prof. Priyanka Kadam, Dr. Zia Saquib

5 Fig:3Dialplan/ extension file for iax.conf THEORITICAL CALCULATIONS FOR BANDWIDTH Following table 1 shows theoretical bandwidth calculation for SIP and IAX protocol for G.711 and GSM codec. These two codecs are selected because they are supported by the asterisk and G.711 gives very good quality of voice and GSM codec is one of the most popular codec. Table 1.Comparison of Bandwidth [5] Protocol Code Bandwidth SIP G Kbps SIP GSM 2184 Kbps IAX G Kbps IAX GSM Kbps RESULTS The following figures 7.1 and 7.2 shows the simultaneous calls to SIP and IAX extensions configured on the Asterisk server, which travel up the stream. With the command ``show channels on the Asterisk console can be displayed in real time, 30 channels, as shown in Figure 4 and figure 5 for SIP and IAX protocal respectively Once we have done 30 simultaneous calls through SIP and IAX protocol, out of them active channel status is indicated by up-stream comment and with the help of show channel command total no of active channels can 51 Prof. Priyanka Kadam, Dr. Zia Saquib

be displayed. To find out bandwidth requirement of SIP and IAX active channels with two different codecs G.711 and GSM we are using erlag s B calculator. Fig: 4.

711 codec. Communication system with SIP protocol and G.711 codec requires highest bandwidth. This is because SIP is having large size of overheads.

6 be displayed. To find out bandwidth requirement of SIP and IAX active channels with two different codecs G.711 and GSM we are using erlag s B calculator. Fig: simultaneous calls through SIP protocol. Fig 5.30 simultaneous calls through IAX protocols. CASE I: Figure 6.3 shows bandwidth requirement of SIP channels by using G.711 codec. Communication system with SIP protocol and G.711 codec requires highest bandwidth. This is because SIP is having large size of overheads. At the same time codec G.711 gives low compressionrate but better voice quality. Hence this combination works well for ideal PBX with relatively low traffic. 52 Prof. Priyanka Kadam, Dr. Zia Saquib

711 codec IAX is ideal for power or compatible with Asterisk IAX whose traffic level is relatively high, since it has a

7 Fig: 6 Bandwidth required for simultaneous 30 calls by using SIP protocol and G.711 codec Case II:-The communication system with G.711 codec IAX is ideal for power or compatible with Asterisk IAX whose traffic level is relatively high, since it has a good voice quality but requires a bandwidth as high. Figure 7 shows simulation results for this case. Fig 7.Bandwidth required for simultaneous 30 calls by using IAX protocol and G.711 codec. 53 Prof. Priyanka Kadam, Dr. Zia Saquib

8 Case III: The communication system configured with SIP and GSM codec is ideal where systems does not support IAX protocol. As IAX is proprietary protocol it is developed to minimize the overhead in signalling. In case of SIP it uses different ports for signalling and media tunnelling is not possible with it. Fig: 9 BW require for 30 calls by using SIP protocal and GSM codec 54 Prof. Priyanka Kadam, Dr. Zia Saquib

Case IV:- The telephone system configured with IAX protocol and GSM codec has the lowest consumption of bandwidth, due to reuse of the headwaters of the network and transport layer and the high

9 Case IV:- The telephone system configured with IAX protocol and GSM codec has the lowest consumption of bandwidth, due to reuse of the headwaters of the network and transport layer and the high compression rate over G.711 GSM. This configuration presents an acceptable voice quality, but at times of high traffic may present distortions. Fig 10: BW require for 30 calls by using IAX protocal and GSM codec CONCLUSION:- Communication system configured with SIP and G.711 codec requires highest bandwidth; because of the large size of the SIP header and the low compression codec G.711. But this combination gives very good quality voice, Which is suitable for PBX with a relatively low level of traffic. Communication system configured with IAX protocol and GSM requires lowest bandwidth, due to reuse of the overheads of the network and transport layer in IAX protocol and the high compression rate over G.711 GSM. This configuration presents an acceptable voice quality, but at times of high traffic may present distortions. 55 Prof. Priyanka Kadam, Dr. Zia Saquib

10 REFERENCES :- [1] Sonaskar, S.D.Giripunje, Low cost IP Private Branch Exchange (PBX), International Journal of Computer Applications, volume 23-no.3, June [2] Pablo Monotoro, Eduardo Casilari, A Comparative Study Of Standards with Asterisk 2009 Fourth International Conference on Digital Telecommunication,2009 IEEE, DOI /ICDT [3] BUR GOODE, SENIOR M EM BER, IEEE Voice Over Internet Protocol (Vo IP) in PROCEEDINGS OF THE IEEE, VOL. 90, NO. 9, SEPTEM BER2002 [4] Pablo Monotoro, Eduardo Casilari, A Comparative Study Of Standards with Asterisk 2009 Fourth International Conference on Digital Telecommunication,2009 IEEE, DOI /ICDT Parra, Octavio J. Salcedo, Neil Orlando DíazMartínez, and Gustavo López Rubio. "Comparison SIP and IAX to Voice Packet Signaling over VOIP." In th IEEE International Conference on Trust, Security andprivacy in Computing and Communications, pp IEEE, [5] Kanika Shah, UtkarshGoel, Mohammed Abdul Qadeer, "Voice - Video Communication on Mobile Phones and PCs' Using Asterisk EPBX",, vol. 00, no., pp , 2012, doi: /csnt [6]RFC SIP: Session Initiation Protocol. J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler. June 2002 [7] Tian, Lu, Nicolas Dailly, QiaoQiao, Jihua Lu, Jiannan Zhang, and Jing Guo. "Study of SIP protocol through VoIP solution of Asterisk." In MobileCongress (GMC), 2011 Global, pp IEEE, [1] Bowman, M., Debray, S. K., and Peterson, L. L Reasoning about naming systems.. [2] Ding, W. and Marchionini, G A Study on Video Browsing Strategies. Technical Report. University of Maryland at College Park. [3] Fröhlich, B. and Plate, J The cubic mouse: a new device for three-dimensional input. In Proceedings of the SIGCHI Conference on Human Factors in Computing Systems [4] Tavel, P Modeling and Simulation Design. AK Peters Ltd. [5] Sannella, M. J Constraint Satisfaction and Debugging for Interactive User Interfaces. Doctoral Thesis. UMI Order Number: UMI Order No. GAX , University of Washington. [6] Forman, G An extensive empirical study of feature selection metrics for text classification. J. Mach. Learn. Res. 3 (Mar. 2003), [7] Brown, L. D., Hua, H., and Gao, C A widget framework for augmented interaction in SCAPE. [8] Y.T. Yu, M.F. Lau, "A comparison of MC/DC, MUMCUT and several other coverage criteria for logical decisions", Journal of Systems and Software, 2005, in press. 56 Prof. Priyanka Kadam, Dr. Zia Saquib

REACTION PAPER 01 TEL 500

REACTION PAPER 01 TEL 500 TEL 500 Session Initiation Protocol Improvement Using Inter-Asterisk exchange Introduction: Within the VoIP network environment, H323, SIP and IAX are three protocols that solve the problem of voice packet