VoIP transmission mechanism based on TCP

Transcription

1 December 2016, 23(6): The Journal of China Universities of Posts and Telecommunications VoIP transmission mechanism based on TCP Shuang Kai, Guan Zhe ( ), Meng Chunzhi State Laboratory of Networking and Switching Technology, Beijing University of Posts and Telecommunications, Beijing , China Abstract As the widespread employment of firewalls on the Internet, user datagram protocol (UDP) based voice over Internet protocol (VoIP) system will be unable to transmit voice data. This paper proposed a novel method to transmit voice data based on transmission control protocol (TCP). The method adopts a disorder TCP transmission strategy, which allows discontinuous data packets in TCP queues read by application layer directly without waiting for the retransmission of lost data packets. A byte stream data boundary identification algorithm based on consistent overhead byte stuffing algorithm is designed to efficiently identify complete voice data packets from disordered TCP packets arrived so as to transmit the data to the audio processing module timely. Then, by implementing the prototype system and testing, we verified that the proposed algorithm can solve the high time delay, jitter and discontinuity problems in standard TCP protocol when transmitting voice data packets, which caused by its error control and retransmission mechanism. We proved that the method proposed in this paper is effective and practical. Keywords voice transmission, bypass firewall, disorder TCP transmission, byte date stream boundary identification 1 Introduction Multimedia communication on the Internet, anytime and anywhere, has become the universal demand of people. Live stream media based on real-time voice has become the main transfer object of the network. Voice transmission technology has become one of the hot points in current research. In the current network, because of the outstanding performance of the UDP in real-time transmission, it has become the main streaming data bearer protocol. As the widespread employment of firewalls on the Internet, which close the UDP port for security and other factors, the normal voice communications between clients are greatly limited. However, the firewall takes a more relaxed open rule for TCP. Hence, TCP can easily pass through the firewall and can be used to transmit voice. Because of the inherent flow control, such as congestion control and error control mechanism, using TCP to Received date: Corresponding author: Guan zhe, guanzhe@bupt.edu.cn DOI: /S (16) transport real-time voice data will cause following problems: 1) When packet loss occurs, the delay and disorder of packets caused by retransmission mechanism render the packets miss their normal playback time, which will cause voice pause, delay and other phenomena, significantly degrade the quality of voice communication. 2) When congestion occurs, TCP will halve the throughput rate by reducing the size of congestion window to accommodate this situation. But the voice data transmission rate is determined by frame rate, and it does not reduce the transmission rate. This will result in the data accumulation in data buffer and affect transmission and smooth playback of voice data, severely lower voice quality. To solve above problems, this paper proposed a TCP-based method to transmit voice data. This method adopts a disorder TCP transmission strategy. By allowing the receiver to read discrete packets, the data acceptance will not be affected by the lost and delayed packets. Accordingly, reduce the delay caused by retransmission. In addition, we also designs a boundary identification

2 Issue 6 Shuang Kai, et al. / VoIP transmission mechanism based on TCP 91 algorithm. It can efficiently identify the boundary of the disordered packets and enable the audio processing module to receive complete voice data. At last, test results from the prototype system conclude the efficiency and practicality of the method. 2 Related work Currently, the TCP-based transmission methods of voice data consist mainly two categories: network-based method and host-based method. The network-based methods mainly refine the performance of voice transmission by improving the quality of the network. In Ref. [1], Braden et al. proposed an extended comprehensive service on the Internet architecture and protocols. The extended service can meet the real-time requirement of various applications. In Refs. [2 3], Kompella et al. introduced the Internet resource reservation protocol with its extensions and supplements, providing QoS-guaranteed real-time voice transmission under routers support. In Ref. [4], Vishwanath et al. used routers with data buffers in them to improve the quality of voice data. In Ref. [5], Charny et al. defined an expedited forwarding per-hop behavior. By configuring the routing nodes on the network, it provides a virtual link with low packet loss rate, low delay and low jitter for real-time data transmission to ensure the performance of real-time voice data. Nevertheless, most of the network-based methods require switches or routers to support high QoS. But these kinds of switches or rooters are rarely used in current network. So their usefulness is limited. However, the host-based method, which can improve quality of voice transmission through the methods provided by host system, has no such limitation. So its universality and adaptability are much better. In Ref. [6], Cocker et al. came up with a strategy to improve the real-time performance of voice transmission by minimizing the data buffer. In Ref. [7], Wu et al. suggested an algorithm to adjust the data transmission rate: the sender can dynamically adjust the transmission rate according to the regular feedback of real-time network packet loss rate from the receiver. In Ref. [8], Park et al. claimed a method that only let the retransmission occur within a limited period to reduce the retransmission delay. In Ref. [9], Németh et al. introduced an optimized TCP-limit protocol which permits dynamical adjustment of congestion window under different situations to reduce the fluctuation of data flow and obliterate delay and jitter to some extent. In Ref. [10], Evensen et al. proposed an optimized TCP called thin-streaming to optimize the transmission strategy. It can improve data transmission performance by adjusting congestion window. Most of above methods rely on the modification of congestion control mechanism. It cannot avoid a significant change to the TCP. Therefore, these methods cannot guarantee the efficiency and stability of transmission. So they are still in the simulation and research stage. To our best knowledge, there is no mature solution to transmission of TCP based VoIP in industry. The most similar work to ours is SCPS-TP, which is a transport protocol for space telecommunication. The best effort mode of SCPS-TP also bears the out-of-sequence segment. However, it never receives the out-of-sequence segment. It just discards the data if the missing segment not appeared when out-of-sequence data queue reaches its size or the timer elapses, which is very different from our scheme. Although it is not the main purpose in this paper, we still compare the two different scheme in the last part of our paper. 3 Voice transmission based on TCP In a network with packet loss, the requirement of orderly data transmission and retransmission mechanism of TCP result in the application layer cannot read disorder packets. This will increase time delay and jitter of the voice data, lead to voice packets miss their normal play time, and cause the delay and pause of calls, dramatically affect the quality of communication. So a new transmission strategy which allows disordered TCP data transmission is badly needed to reduce the delay of retransmission and guarantee the quality of voice in the network with low loss packet rate. Moreover, because the TCP is a byte stream oriented protocol, it is impossible to distinguish the boundary information of the user media data packets. Thus, allowing disordered TCP transmission strategy may lead to packet boundary information lost and the receiver cannot restore the original voice date. This will make the decoder not work. Therefore, it is also desirable to design a byte stream data boundary identification algorithm to identify the received voice packets, and then, delivery them to the voice audio processing module timely.

3 92 The Journal of China Universities of Posts and Telecommunications Overview of the overall architecture The network architecture of TCP based voice transmission method proposed in this paper is shown in Fig. 1. The reformed voice terminals are behind the firewall, the server s firewall blocks UDP flows and allows TCP flows to get through. The system completes the call process by implementing the disorder TCP transmission strategy and byte stream data boundary identification algorithm. Fig. 1 Overall architecture The software architecture of voice terminal is shown in Fig. 2. The disordered TCP transmission strategy is implemented by modifying the TCP protocol stack. The intermediate layer unit contains data coding strategy, boundary identification strategy and data decoding strategy. The boundary identification of byte flow is based on consistent overhead byte stuffing algorithm. Fig. 2 Software architecture of voice terminal 3.2 Disorder TCP transmission strategy The disorder TCP transmission strategy enables the application layer to read latter data packets in a serial date sequence without having to receive the preorder data packets in the same sequence, so as to avoid pause and time delay in a call process. The strategy does not modify any interface of TCP. It only adds a new socket option: SO_UNORDERED, to open the disorder TCP transmission strategy. The TCP with such a strategy allows the application layer to read the data whenever new data packets arrive, skipping those not arrived packets. The application layer use the same read() function to read data, but the first byte of the data returned by read() function is not necessarily follows the last byte of date returned by the last call to read() function. However, the read() function will transfer all the data to the application layer at least once. And the disorder TCP transmission strategy will add a 4B header at the beginning of the data to identify the offset of the first byte of the data in original byte stream. With the header, the receiver can use the received data to restore the original data. Fig. 3 shows the comparison of receiving behaviors of standard TCP and disorder TCP transmission strategy. In Fig. 3, the first packet arrives in right order, the second arrives out of sequence and the third is the missing packet before the second packet. When using standard TCP to transfer data, the first packet can be read directly by the application layer, the second packet stays in the queue until the missing data the third packet arrives, then the second data and third data can be read by the application layer in sequence. But when using the disorder TCP transmission strategy, the first packet can be read directly, and the second packet also can be read immediately as disorder packet without waiting for the missing data. (a) Standard TCP transmission

4 Issue 6 Shuang Kai, et al. / VoIP transmission mechanism based on TCP 93 (b) Disorder TCP transmission Fig. 3 Receive behavior of disorder TCP transmission 3.3 Byte stream data boundary identification algorithm The byte stream data boundary identification algorithm is implemented by the intermediate layer unit, which contains data coding strategy, data decoding strategy and boundary identification strategy. When sending the data, the intermediate layer firstly uses consistent overhead byte stuffing algorithm to encode data, removing 0x00 B in the data sequence. Then, add a 0x00 B at the beginning and the end of each packet as boundary flag. When receiving data, the receiver can identify a complete voice packet by recognizing the continuous data segment starts and ends with 0x Data coding strategy Data coding strategy is used to encode voice data. It serves as the foundation of boundary identification. The strategy uses consistent overhead byte stuffing algorithm to encode data. The result of this strategy is that all 0x00 B are removed from the data sequence. After that, a 0x00 B is added to both the head and the end of the data sequence to represent a complete data packet. The specific steps of encoding are shown in Fig. 4. 1) Traverse the data packet and find out all positions that 0x00 B present. Then based on these 0x00 B, the original data packet is divided into several data blocks with a 0x00 B tail for each. The length of the data blocks depends on the positions of the 0x00 B in the packet. So a data block may contain only one byte or several bytes. 2) Use data coding strategy to encode data blocks above. 3) Assemble the coded data blocks and reconstruct a new data packet. 4) Add a 0x00 B respectively at the head and the end of the new data packet as identification of the packet. Fig. 4 Data coding steps Boundary identification strategy Boundary identification strategy is used to identify the boundary of the disorder TCP data packets. The strategy guarantees the received complete voice data packets can be delivered to audio processing module timely so as to improve the quality of voice calls. Fig. 5 shows all three possible situations that boundary identification strategy has to tackle with: 1) In situation 1, the sender sends 3 consecutive voice data packets which have been encoded by intermediate layer. There are no routers and other equipment in the

5 94 The Journal of China Universities of Posts and Telecommunications 2016 network, and the packet 2 is lost in transmission. Thus, only packet 1 and packet 3 can be received by intermediate layer under this circumstance. By identifying the 0x00 B, the receiver gets two packets with 0x00 B header and 0x00 B tail. Then the two packets are decoded and delivered to audio processing module to play, without waiting for the missing packet. As a result, the time delay and pause in voice calls can be avoided and the loss of a small amount of voice data will not significantly affect the quality of calls. 2) In situation 2, the sender sends three consecutive data packets as well. But what different from situation 1 is that the three packets are fragmented and reassembled by routers and other network equipment and are transformed into two packets. One of the new packets consists of packet 1 and the front part of packet 2, the other consists of packet 3 and the remaining part of packet 2. When receiver receives the first reassembled packet, it can restore the original data segment by identifying the 0x00 B boundary and decoding the consecutive data sequence through intermediate layer. Then, the remainder of the first packets is attached to the next packet. Repeat the process above, we can get all three original packets. 3) Situation 3 is similar to situation 2, but the first reassembled packet is lost. The receiver only received the second reassembled packet. As a result, only packet 3 is restored after boundary identification and decoding. Therefore, only packet 3 is passed to the application layer. This ensures the voice will be played continuously without pause and high time delay. Fig. 5 Situations in boundary identification Data decoding strategy The data decoding strategy takes the output of the boundary identification strategy as input. The complete original voice packet can be restored by removing the marker byte data at the head and the end of the data sequence. The process of the decoding is the inverse of coding. So it is not necessary to discuss any detail here. 4 Prototype system and test results 4.1 The prototype system This paper modifies the TCP protocol stack of Linux system to implement the intermediate layer unit. At the same time, the paper also modifies the Java-based scientific instruction processor (SIP) terminal. By replacing UDP with TCP as its lower media transmission protocol, the terminal now includes the intermediate layer we have described above. The network architecture of test system is shown in Fig. 1. In the network, two voice terminals are running on two Linux servers, the firewalls of which are set to block UDP flow and enable TCP flow to get through. We use dummynet to control the network latency and packet loss rate and set up a network environment with low packet loss rate (0~2%). To evaluate the validity of the method, we analyze the time delay from one terminal to another and the data processing performance of the terminal. 4.2 Voice quality analysis For the quality of the VoIP system, an import factor is whether the voice packets can be played at their right playing time. In VoIP calls, the voice data has certain tolerance on packet loss. When 90% or more voice packets can be played normally, the users communication will not be influenced. But when less than 80% of the voice packets can be played normally, the communication will not be carried out. That means 80% of the voice data must be received and discerned by the voice terminal before the right playback time. The experiment tests the voice quality under various

6 Issue 6 Shuang Kai, et al. / VoIP transmission mechanism based on TCP 95 conditions where voice is carried by UDP, standard TCP and modified TCP, which allows disorder transmission, under different packet loss rate. The test result is shown in Fig. 6. environment, we compared the performance of the two models under different package loss rate. The result of the experiment is showed in Fig. 7. It suggests that disorder TCP outperforms the best effort mode of SCPS-TP. When the package loss reaches 2%, the SCPS-TP missed near 5% of the voice package, while disorder TCP can normally paly 98% of the VoIP package. The result really makes sense because SCPS-TP does not deliver the voice package to application layer if the missing data never reaches the receiver. On the contrary, our disorder TCP model passes the out-of-sequence packages to application layer immediately without waiting for the missing data. Fig. 6 Proportion of normally played voice packets The result suggests that when voice is carried by UDP, because UDP is not required to guarantee the reliability and real-time of the data, the vast majority of voice packets can be played at their playing time. Only the lost packets cannot be sent to opposite terminal to play due to the UDP does not provide retransmission mechanism, but this will not significantly impact the quality of voice calls. And when packet loss rate rises to 2%, although the proportion of correctly played voice packets has a bit decline, 95% of the voice packets are still can be played at right time and this will not affect voice quality. When voice is carried by standard TCP, because the TCP is required to guarantee the reliability of voice data, retransmission mechanism will be used to resolve the packet loss problem. So, it will introduce more extra time. Moreover, with the increasing of packet loss rate, the proportion of normally received and played voice packets rapidly decrease. When the packet loss rate is 0.7%, there are only less than 90% of the voice packets can be played normally. When the rate increases to 1%, it is almost unable to complete communication. The method proposed in this paper is based on TCP protocol and allows application layer to read data directly from the disorder TCP data sequence and timely deliver the received integrated packets to upper processing module. When packet loss rate is lower than 1%, the effect of the method is close to that of UDP. And when the rate increases to 2%, there are more than 90% of the voice packets still can be played normally. This guarantees the normal communication between two clients. To compare with SCPS-TP, we performed another comparative experiment. Giving the same network Fig. 7 Contrast between SCPS-TP and disorder TCP 4.3 Performance of voice terminal The method proposed in this paper will increase time cost because disorder TCP transmission strategy and boundary identification strategy have to spend extra time in data processing. In Fig. 8, we compare the CUP processing time of standard TCP transmission and disorder TCP transmission proposed in this paper. Each column of the histogram has a dark part and light part, the dark part represents the processing time of the kernel of TCP protocol, the light part is the time used to process voice data in voice terminal. The processing time of traditional TCP is used as unit time in Fig. 8. We can see that in the process of using the method proposed in this paper, the kernel s overhead of disorder TCP protocol stack is basically consistent with that of the standard TCP transmission, the disorder TCP transmission strategy nearly does not affect the kernel s processing time of protocol stack. However, due to the series process of data coding, boundary identification and data decoding in intermediate layer unit, the processing time of voice terminal will increase (2~5 times more than the time cost

7 96 The Journal of China Universities of Posts and Telecommunications 2016 of standard TCP). But the additional time cost will not change a lot in different packet loss rate, and the cost is within reasonable range when compared with the overhead of timeout retransmission mechanism (almost hundreds times more than the processing time of TCP protocol stack). So the method has a very high availability. more convincing. Actually, our method has been accepted by a known communication operator and is under network test now. Acknowledgements This work was supported by the National Natural Science Foundation of China: Research on Differentially Private Frequent Pattern Mining, the National Natural Science Foundation of China: Research on Optical Transport Network Integrated Protection Strategy and Method for High-Speed Railway References Fig. 8 Contrast between standard TCP transmission and disorder TCP transmission on CPU time cost 5 Conclusions In the current Internet environment, the voice packets carried by UDP always encounters such situations, like abandoned by routers and cannot pass through the firewall, which affect the normal communication in UDP-based VoIP system. In this paper, we analyzed the characteristics of voice transmission based on TCP and proposed a novel method to transport voice data using modified TCP. In the method, we presented a disorder TCP transmission strategy and designed a byte stream data boundary identification algorithm to solve the high time delay, jitter and discontinuity problems which introduced by the error control and retransmission mechanism in standard TCP when transmitting voice data. The verification through prototype system demonstrates the method proposed in this paper is effective and practical. Moreover, the comparative experiment with SCPS-TP makes our method 1. Braden R, Clark D, Shenker S. Integrated services in the Internet architecture: an overview. IETF RFC 1633, Kompella K, Lang J P. Procedures for modifying the resource reservation protocol (RSVP). IETF RFC 3936, Yoo H, Yoon J, Lee S, et al. A mobile reservation protocol for video streaming. Proceedings of the 10th International Conference on Advanced Communication Technology (ICACT 08): Vol 3, Feb 17 20, 2008, Phoenix Park, Republic of Korea. Piscataway, NJ, USA: IEEE, 2008: Vishwanath A, Sivaraman V, Rouskas G N. Anomalous loss performance for mixed real-time and TCP traffic in routers with very small buffers. IEEE/ACM Transactions on Networking, 2011, 19(4): Charny A, Bennet J C R, Benson K, et al. Supplemental information for the new definition of the EF PHB (expedited forwarding per-hop behavior). IETF RFC 3247, Cocker E, Ghazzi F, Speidel U, et al. Measurement of buffer requirement trends for real time traffic over TCP. Proceedings of the IEEE 15th International Conference on High Performance Switching and Routing (HPSR 14), Jul 1 4, 2014, Vancouver, Canada. Piscataway, NJ, USA: IEEE, 2014: Wu D, Hou Y T, Zhu W, et al. On end-to-end architecture for transporting MPEG-4 video over the Internet. IEEE Transactions on Circuits and Systems for Video Technology, 2000, 10(6): Park C, Kim H. Short-term reliable protocol for low latency video transmission. Proceedings of the 2014 International Conference on Computational Science and Computational Intelligence (CSCI 14): Vol 2, Mar 10 13, 2014, Las Vegas, NV, USA. Piscataway, NJ, USA: IEEE, 2014, 2: Németh F, Molnár S, Tarján P, et al. TCP limit: a streaming friendly transport protocol. Proceedings of the 4th EURO-NGI Conference on Next Generation Internet Networks (NGI 08), Apr 28 30, 2008, Krakow, Poland. Piscataway, NJ, USA: IEEE, 2008: Evensen K, Petlund A, Griwodz C, et al. Redundant bundling in TCP to reduce perceived latency for time-dependent thin streams. IEEE Communications Letters, 2008, 12(4): (Editor: Wang Xuying)