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 signaling. In this paper, SIP and IAX are compared based on use of bandwidth as a critical factor. Basic requirements for an audio application conversational, Possible to use audio codec s, VoIP conversations overhead caused by communication protocols involved in, Analysis of the format of SIP and IAX frames with its own package are also taken into account by applying them for theoretical calculations of bandwidth consumption. VoIP Protocols: There are two types of VoIP protocols. Centralized and distributed. The above mentioned protocols are distributed type which allow network intelligence. It is distributed between the call control devices, VoIP gateways, IP phones, media servers or any device that can initiate and terminate a VoIP communication. VoIP Architecture Models: Application Presentation Session Transport Network Data link Physical Asterisk G.729/G711/GSM/Speex H323/SIP/MGCP/IAX UDP/RTP/SRTP IP/CBWFQ/WRED/IP Precedence/Diffserv Frame-Relay/ATM/PPP/Ethernet Ethernet/V.35/RS-232/Xdsl Architectural model of VoIP protocols within the OSI model SIP (Session Initiation Protocol) Protocol: It is a session layer protocol developed and standardized by the IETF (Internet Engineering Task Force). It is based on the mechanism of request and response to initiate a communication session, allowing sessions and video data between two endpoints. After establishing a session, average flow of transfer is handled by RTP protocol. IAX (Inter Asterisk exchange) Protocol: The two main objectives of the project IAX derived from experience with VoIP protocols like SIP and MGCP for control and RTP media stream such as minimizing the use of bandwidth with specific emphasis on voice conversations, provide NAT transparency and structure, signaling and media flow transfer to IAX. P a g e 1 4
IAX uses UDP port 4569 to communicate all the packages. Trunked IAX protocol allows the use a single header for the passage of several calls. However, so far IAX trunked mode can only be enabled between two Asterisk servers. Parameters Required For The Analysis Of The Use Of The Bandwidth: Codec Sample Interval (CSI) (ms) Codec Sample Size (CSS) (Bytes) Codec Bit Rate (CBR) (kbps): is calculated by following expression: CBR = Voice Payload Size (VPS) Packets per Second (PPS): is calculated by the following expression: PPS = Total Packet Size (TPS) (bytes): is given by the following expression: TPS = L2 + IP + UDP +L5 + VPS Where, L2 = size of the protocol header data link layer L5 = size of the session protocol header Bandwidth (BW) (Kbps): the bandwidth required for n conversations full duplex BW n = BW * n * 2 BW Calculation Protocol SIP codec G.711 Input data: CSI = 10 ms, BW30calls = 87.2 Kbps x 30 x2 = 5232 CSS = 80 bytes, VPS = 160 bytes Kbps (more than 5 Mbps) BW Calculation Protocol SIP GSM codec Input data: CSI = 10 ms, BW30calls = 36.4 Kbps x 30 x 2 =2184 CSS = 80 bytes, VPS = 160 bytes Kbps (near 2 Mbps) IAX Protocol G.711 codec Input data: CSI = ms, BW30calls = 84 Kbps + 65.6 Kbps x 29 CSS = 80 bytes, VPS = 160 bytes, =1986.4 Kbps (near 2 Mbps) CBR = 64 Kbps, VPS (ms) = 20 ms, PPS = 50 Input data: CSI = 20 ms, CSS = 33 bytes, VPS = 33 bytes, CBR = 13.2 Kbps, VPS (ms) = 20 ms, PPS = 50 H.IAX Protocol codec GSM Results according applied protocol BW30calls = 33.2 Kbps + 14.8 Kbps x 29 = 462.4 Kbps (Less than 512 kbps) P a g e 2 4
Traffic Generation: Call Simulator: The simplest way to make calls is through configuration files. For this, you need to configure a file extension *.call by call. Thus, 30 files must be configured to simulate the same number of words. Channel, context, extension and priority are the basic parameters. The files must be located in the /var/spool/asterisk/outgoing, but must be created in a different location. As soon as they are moved to /var/spool/asterisk/outgoing, the center recognizes and, according to the parameters described in them, the call is made. General Procedure: For this analysis we perform the following steps: Vyatta system installation on computers that act as routers. Configuring interfaces and static routes on the Vyatta. Installing CentOS 5.2 operating system on servers Asterisk Asterisk 1 and 2. Installing Asterisk PBX on the servers 1 and 2. Installing Wireshark and Unsniff sniffer on the machine. SIP settings 30 extensions (101-130) and 30 IAX extensions (201-230) in the Asterisk server 2 (Editing sip.conf files and iax.conf, respectively) Dial plan configuration on the Asterisk server 2, so that when receiving a call to run a default recording that will simulate the flow of media (Edit the extensions.conf file) 60 files were created with. Call in the location / var / spool / asterisk / tmp on the Asterisk server 1, 30 for SIP (101.call - 130.call) and 30 for IAX (201.call - 230.call) with which to generate calls from the server 1 to 2. Dial plan configuration on server 1, creating a context, extension and symbolic priority from where they will launch the Asterisk server calls 2. Asterisk software to run on both PBXs. SIP Analysis with Wireshark It starts a capture in Wireshark by checking the appropriate choice of the network interface through which data must be captured. Should move, not copy-files. Call (101.call - 130.call) to / var / spool / asterisk / outgoing, must keep a backup of those files for later use. With the command "show channels" on the Asterisk console can be displayed in real time, 30 channels. When the call is finished, stops the capture of Wireshark and saved. P a g e 3 4
Conclusions: Protocol Codec BW Theor BW Real SIP G.711 5232 Kbps 5131 Kbps SIP GSM 2184 Kbps 2087 Kbps IAX G.711 1986.4 Kbps 2166 Kbps IAX GSM 462.4 Kbps 552 Kbps Theoretical results for the use of bw The telephone system configured with SIP and G.711 codec has the highest consumption of bandwidth; this is due in large part to the size of the SIP header and the low rate compression codec G.711. This configuration, however, presents a very good quality voice, making it ideal for PBX with a relatively low level of traffic. 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. The telephone system configured with G.711 codec IAX and 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. The telephone system configured with SIP and GSM codec is ideal for plants that do not support IAX, presenting the same advantages that the previous configuration. P a g e 4 4