Experimental Study of SIP and Customized Satellite SIP (CSS) Protocol over Satellite Network

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1 Experimental Study of SIP and Customized Satellite SIP (CSS) Protocol over Satellite Network Nidhi Raja 1, Raju Das 2, Sudhir Agrawal 3 1 LJIET, Ahmedabad 2 Scientist, Space Applications Centre ISRO, Ahmedabad 3 Head, DCTD, Space Applications Centre ISRO, Ahmedabad Abstract IP datagram became de-facto standard for all most all types of communication. Satellite communication is power and bandwidth limited. IP communication protocol (like VoIP) is not optimized for satellite network. Performance is affected by mainly three factors - long delay, limited bandwidth and channel error. To get optimum performance of VoIP standards over satellite network we need to customize or modify the existing protocols to suits the channel characteristics of satellite network. The paper presents an analysis and comparison of SIP protocol and new Designed Protocol in terms of call loss, call setup time and bandwidth utilization. Index Terms VoIP, SIP Protocol, Satellite Network I. INTRODUCTION Now a day different Multimedia applications are widely used over Internet. It gives an attractive solution for voice/data integration in public and private networks. The satellite links have capacities to carry data packets. The satellite networks have global coverage and also reach to remote areas. Originally VoIP standards are designed for terrestrial link which may not give optimum performance when apply over satellite network. So for this some modifications are required to suits satellite link conditions. Different VoIP protocols are defined here. H.323 defines the technical requirements for multimedia communications in local area networks, enterprise area networks, metropolitan area networks, intranets, internets etc. Now a day the SIP taken place of H.323as the SIP is simpler than H.323 in terms of developing and supporting software. H.323 contains a set of standards for multimedia data transmission without a guarantee of the quality of service (QoS). SIP is used to establish and terminate the user session. SIP is a protocol with less complexity and more flexibility. SIP makes effective use of Session Description Protocol (SDP). The end node is informed by the SDP that encoder/decoder handles which type of sessions. Most VoIP applications are real-time. Voice packets are sent from source to destination with Real-time Transport Protocol (RTP). It is controlled by Real Time Control Protocol (RTCP). The end to end delivery is provided by RTP with a time stamp, payload type identification and sequence numbering. It runs on User Datagram Protocol (UDP). The main function of RTCP is to provide feedback on the quality of the data distribution and informs RTP about any feature which requires change. Normally disaster occurs without prior knowledge or information. First victim to the disaster is the terrestrial communication system. And the most urgent requirement of post disaster is the restoration of communication system. The effective solution in this situation would be establishing a satellite link between affected areas to the safer world. Installation of Satellite nodes needs laser time and can cover larger area compare to terrestrial network. The last-mile connectivity (i.e. the connection between user devices and satellite node) can be offered to the users (the administrative personal as well as the victims) by introducing many state-ofart terrestrial technologies. The proposed disaster network will be using IP convergence to carry media traffics from multiple services. While we using VoIP over the satellite network, the performance is affected by mainly three factors - long delay, limited bandwidth and channel error. To get optimum performance of VoIP standards over satellite network we need to customize or modify the existing protocols (by performing different operations like header compression or SIP signaling compression) to suits the channel characteristics of satellite network. The optimization of the signaling (signaling compression) for better performance in satellite network in terms of call loss, call establishment time and bandwidth utilization is done by introducing the new Protocol between the proxies servers transmit over the satellite network. The paper is organized as follows. Section II provides the system description. In section III CSS Protocol design is described. Experimental results are given in section IV followed by conclusion. IJIRT INTERNATIONAL JOURNAL OF INNOVATIVE RESEARCH IN TECHNOLOGY 200

II. SYSTEM DESCRIPTION The Fig.1 shows the satellite based disaster network. The network operates in star mode. The network is managed by centralized Network Management System at HUB.

2 II. SYSTEM DESCRIPTION The Fig.1 shows the satellite based disaster network. The network operates in star mode. The network is managed by centralized Network Management System at HUB. The terrestrial gateway is located at HUB to route calls to external public network (mobile/land line). All VSAT terminals are installed at different remote location and every satellite terminal will be equipped with SIP Proxy to provide user interface for making call. The user can make VoIP calls to local users (connected with same SIP Proxy) or to the remote users (connected with other remote VSAT). A user can also dial any terrestrial public network subscribers through Gateways which are installed at HUB. connection splitting mechanism is used for communication. The proposed Customized Satellite SIP (CSS) Protocol will run, between the proxies, over satellite network to get better performance. Fig. 2 Proposed System Diagram User will always start to communicate with standard SIP protocol. When a new request comes to a proxy, it extracts the required information from the request message packet and generates CSS request packet (all CSS formats are given in next section) to transmit to the destination end proxy. Similarly the receiving proxy will regenerate the SIP request packet from the received CSS message and sends to the called user. III. CSS MESSAGE FORMAT All CSS message format is given here. Fig. 1 Satellite based Disaster Network Fig. 3 CSS Request message format (17 bytes) The system with three servers- SIP Proxy Server, DHCP Server & Web Server is being developed to provide last mile connectivity for users (mobile or fixed connection) to make voice/video call through satellite especially during disaster. DHCP, SIP Proxy and Web servers need to customize with some specific features and implement on single embedded system to and meet specific requirement of disaster. One of the important parts of it is SIP Proxy server, through which a conversation session is established. SIP is widely used multimedia standard for audio/video conferencing. The SIP protocol originally designed for terrestrial IP network and which may not be efficient for satellite network as it is (mainly for path delay, channel error rate and bandwidth utilization). So there is a need for development of SIP proxy server which will route all VoIP traffic efficiently through the satellite network The Fig.2 shows the how the proposed system will work. All calls must be routed though proxies and the Fig. 4 Protocol analysis in Wireshark Message Type ID (1 Byte): Type of the message. o Invite (11), o 180 Ringing (13), o 200 OK (14), o ACK (15), o Bye (16) o Call cancel (17) Destination Call ID (1 Byte): Assigned by destination proxy and this id will be used for further communication. Initially in Invite packet the destination call ID is 0. The number will assign the destination proxy after receiving a CSS invite. IJIRT INTERNATIONAL JOURNAL OF INNOVATIVE RESEARCH IN TECHNOLOGY 201

umber of Packet loss June 2015 IJIRT Volume 2 Issue 1 ISSN: 2349-6002 Source Call ID (1 Byte): Assigned by calling proxy on receiving a new SIP request from user.

3 umber of Packet loss June 2015 IJIRT Volume 2 Issue 1 ISSN: Source Call ID (1 Byte): Assigned by calling proxy on receiving a new SIP request from user. This id will be used for further communication. Destination Call Number (6 Byte): Defines the called party number. The number is in BCD format (e.g will be formatted as 0x00 0x09 0x87 0x65 0x43 0x21) Source Call Number (6 Byte): Defines the calling party number. Checksum (2 Byte): To check the data integrity at both ends. All other message formats (OK, cancel, ringing, bye) are similar with 5 bytes length Packet Loss Paclet loss(sip) Packet loss Fig. 5 CSS message format (5 bytes) Fig. 7 BER vs. Packet loss for SIP and CSS Protocol IV. Fig. 6 Protocol analysis in Wireshark EXPERIMENTAL RESULTS Simulation has been done to compare performance of CSS protocol and SIP Protocol over satellite network in terms of bandwidth utilization, call loss and call setup time. Communication between systems is through emulator to set delay and different BER rate. A. Packet loss The satellite network is error prone and bandwidth limited. While using the larger size signaling frame format like in SIP over the satellite network, the number of call loss is increased. By using the CSS Protocol it reduces the large number of packet loss compare to SIP protocol. Table 1. Number of Packet Loss for SIP and CSS Protocol at different BER rate Total number of Packet Packet loss of SIP Protocol Packet Loss of CSS Protocol * * * The SIP message formats are of larger in length that requires more satellite bandwidth and also suffers more packet losses over satellite link. From the Fig.7 it concludes that CSS Protocol with necessary fields is generated at proxy largely reduces the packet loss which means less number of call loss compare to the SIP Protocol. B. mechanism To handle packet loss in the transmission retransmission is done. More number of retransmission is needed when channel error rate increases. If the response is not be received at the transmitter side within 600 ms (500 ms is RTT and 100 ms is processing time), then the retransmission of the packet is done. Different Numbers of retransmissions are needed at different BER to get 100% success response is shown in the Table 2. Table 2. Number of retransmission needed for SIP and CSS Protocol at different BER Bit Error Rate No of (CSS No of (SIP IJIRT INTERNATIONAL JOURNAL OF INNOVATIVE RESEARCH IN TECHNOLOGY 202

4 Time in Seconds E-10 June 2015 IJIRT Volume 2 Issue 1 ISSN: The call setup time is direct proportion to number of retransmission. By using CSS Protocol between the proxies the less number of retransmissions are needed compared to SIP Protocol. So it reduces the call setup time of CSS Protocol compare to SIP Protocol. C. Multiple Packet Sending Mechanism Chart Title retransmissio n(sip Another method, we have implemented here, to mitigate packet loss is by sending redundant packets. The receiver also sends multiple response packets. More number of redundant packets is required when the channel condition goes bad. Retransmissio n(css Table 3. Call setup time for Multiple Packet sending and at different BER Bit Error Rate (SIP (CSS Multiple sending (CSS Reduction in call setup time by using multiple sending mechanism compare to retransmission of SIP Protocol and CSS Protocol is shown in Table 3. By retransmitting packet, after a receive timeout, we get the 100% success rate, but the average call setup time is increased. The mechanism which gives 100% success response and also reduces the call setup time is sending the multiple packets at a time. Fig. 8 Call setup time for SIP and CSS protocol with retransmission and multiple sending mechanism V. CONCLUSION To get optimum performance of VoIP standards over satellite network need to customize or modify the existing protocols to suits the channel characteristics of satellite network like limited bandwidth, channel error which is the reason of increasing call drop and call setup time. For this new Protocol CSS has been designed to improve the performance over the satellite network. The performance of the CSS Protocol is far better than the SIP Protocol in terms of bandwidth utilization, call drop and call setup time. CSS Protocol Reduces the number of call drop and number of retransmission at different BER compare to the original SIP Protocol. is performed to handle the packet loss in communication link. By retransmitting packet we get the 100% success response but the call setup time will be increased. Another solution to get 100% success response is by sending the multiple packets at a time. It reduces the call setup time compare to previous method of retransmission. ACKNOWLEDGMENT This method consumes more bandwidth compare to previous one, but gives better QoS in terms of call establishment time. As the CSS protocol frame formats are of smaller size (17 and 5 bytes), so it needs little more extra bandwidth compare to retransmission methods, but gives better QoS parameter as we do not require to retransmission the same packet after the acknowledgement timeout. Authors are thankful to Shri Kaushik Parikh, Dy Director, SNAA/SAC/ISRO and Shri Virender Kumar, GH, SSTG/SNAA for ISRO for his continuous guidance, encouragement and support. The authors sincerely appreciate the critical evaluation and constructive suggestions provided by the reviewers: Prof. Gayatri Pandi (Jain), Jignesh Vania. REFERENCES 1] M. Ali, L. Liang, Z. Sun and H. Cruickshank, "Evaluation of SIP Signaling and QoS for VoIP over Satellite Networks", IEEE International Conference on Communication, ICC, June 2009, ISSN: Page(s): ] Dong-Yeop Hwang, Ji Hong Park, Seung-WhaYoo, Ki- Hyung Kim, "A Window-Based Overload Control Considering the Number of Confirmation Massages for IJIRT INTERNATIONAL JOURNAL OF INNOVATIVE RESEARCH IN TECHNOLOGY 203

5 SIP Server", International Conference on Ubiquitous and Future Network, ICUFN, July ISSN: Page(s): ] Weihai Li, Yan Ma, Qing Ma, Xiaohong Huang, "Dynamic Dictionary Design for SIP Signaling Compression", WRI Eorld Congress on Computer Science and Information Engineering, March-April Page(s): ] Tao Wen, Dongqing Zhang, Quan Guo, A New SIP Compression Mechanism with Pretreatment Based on SIGCOMP in IMS, International Conference on Internet Technology and Applications, ITAP, August 2011.Page(s):1-6. 5] IJIRT INTERNATIONAL JOURNAL OF INNOVATIVE RESEARCH IN TECHNOLOGY 204

Transporting Voice by Using IP

Transporting Voice by Using IP National Chi Nan University Quincy Wu Email: solomon@ipv6.club.tw 1 Outline Introduction Voice over IP RTP & SIP Conclusion 2 Digital Circuit Technology Developed by telephone