Evaluation of SIP-based VOIP in Heterogeneous Networks Mohamed Khedr*, Onsy Abd El Aleem** Mohamed Mahmoued Selim*** *Arab Academy for Science and Technology, Alexandria, Egypt ** Alexandria University, Electrical Department *** Telecom Egypt Khedr@aast.edu, onsy20@hotmail.com, moh.slim@gmail.com Abstract Session Initiation Protocol (SIP) provides advanced signaling and control functionality for a wide variety of multimedia services (audio and video). This paper investigates the effect of end-to-end delay, packet loss and jitter on SIP-based VOIP calls. The investigation is carried out on various media types (10 base-t, 100 base-tx, WLAN) with various SIP registrar (inbound and outbound). To satisfy QoS requirements of VOIP, end-to-end delay should be about 50-100ms. End-to-end delay larger than 300ms is unacceptable to most callers. We propose three different scenarios to study this situation for each type of SIP registrar (inbound and outbound). Experimental results are carried out using about eleven different (SIP-based) IP telephony calls that were carried out on each of the above mentioned scenarios. In each scenario the effect of the media type on delay, packet loss and jitter are considered and also the effect of the registrar location on the overall delay and jitter are investigated, Cross correlation and auto correlation were used to evaluate the delay also The quality of the transmitted speech is subjectively tested by a number of listeners judgments which we call the voice mean opinion score (MOS) I. INTRODUCTION Generally Voice over IP (VOIP) system uses the Internet Protocol (IP) to transmit voice as packets over an IP network. Using VOIP protocols, voice communications can be achieved on any IP network regardless it is Internet, Intranets or Local Area Networks (LAN). In a VOIP enabled network, the voice signal is digitized, compressed and converted to IP packets and then transmitted over the IP network. VOIP signaling protocols are used to set up and tear down calls, carry information required to locate users and negotiate capabilities. The key benefits of Internet telephony (voice over IP) are the very low cost; the integration of data, voice and video on one network, the new services created on the converged network and simplified management of end user and terminals. Although the evolution of Internet telephony is experiencing significant growth due to its low-price for long distance calls, Internet telephony is also experiencing significant challenges such as good reception quality, delay, jitter, packet loss, Band Width and other parameters known as the quality of service (QoS) [1]. The issue is how to guarantee that packet traffic for a voice or other time sensitive media will not be delayed or dropped due to interference from other lower priority traffic in the network. Quality of Service for voice over IP network is a critical issue because it is real time and delay sensitive application. Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for establishing VOIP connections (multimedia session). SIP is an application layer protocol for creating, modifying and terminating multimedia sessions with one or more participants. Borrowing from Internet protocols, such as HTTP (hyper text transfer protocol), SIP is text-encoded and highly extensible [2]. SIP is an application-layer control (signaling) protocol for setting, modifying, and terminating sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephony and multimedia distribution. Members in a session can communicate via multicast, mesh of unicast, or a combination of these. SIP uses the Real Time Transport Protocol (RTP) for transmitting audio and other time sensitive data and uses the Session Description Protocol (SDP) for describing multimedia sessions [1]. RTP is used over UDP to send audio and video data [3, 10]. RTCP (Real-time Transport Control Protocol) is used along with RTP, to provide some QoS information. For a point-topoint case the RTCP information can provide delay, jitter and loss information in each direction to both participants. Since SIP targets time-sensitive applications, several delays can be considered to assess the quality of service of this protocol. In this paper, we focus on studying the effect of the location of the SIP registrar server and the effect of the media used between user agent clients (UAC) and registrar in the end to end delay, packet loss and jitter as they impact the over all QoS of the system.end-to-end delay does not affect voice quality directly but it is an important factor in determining whether subscribers can interact normally in a conversation taking place over an IP network or may experience near like blocking situation. Reasonable delay figure should be about 50-100ms. End-to-end delay larger than 300ms is unacceptable to most callers. Jitter ad packet loss on the other hand directly affects the quality of voice as it deals with the variations in delay of the received voice. The rest of this paper is organized as follows. Section II briefly explains SIP and discusses related work that has been done to investigate the End-to-End delay. Section III presents the process that we followed to evaluate and measure the Endto-End delay and jitter of SIP calls. Experimental results for the delay, jitter, packet loss and distortion are explained in Section IV. Section V discussed the listeners' opinion and judgment by means of Mean opinion score and Conclusion and future work as section VI.
II. SIP AND RELATED WORK Four logical types of entities participate in SIP-based VOIP calls: user agents(client 'UAC' or server 'UAS'), registrars, proxy and redirect servers. User agents initiate requests and they are also the final destination. Internet telephones and conferencing software are examples of user agents. Registrars keep track of users within their assigned network domain (e.g., all users with identifiers x@macrosoft.com register with the registrar in the macrosoft.com domain). Proxy servers are application-layer routers that forward SIP requests and responses. Redirect servers receive requests and then return the location of another SIP user agent or server where the user might be found [1]. It is quite common to find proxy, redirect, and registrar servers implemented within the same program. In a typical SIP session, SIP messages originating at a user agent traverse one or more SIP proxy servers and then reach one or more SIP user agents. However, SIP user agents can also communicate directly with each other [1]. Indeed, it is common that only the first request exchange travels along a chain of proxies, with all subsequent requests exchanged directly between the two user agents. In the example shown in figure 1, no outbound proxy is used. Alice, using the IP phone with the domain name a.wonderland.com, wants to call Bob in the macrosoft.com domain. Her user agent routes the request to the designated server for macrosoft.com, where Bob has registered his SIP phone as b.macrosoft.com. Bob is on vacation and has forwarded his calls to Carol in the same domain, who has registered at the machine c.macrosoft.com. The proxy server thus forwards the request there, replacing the request URI with carol@c.macrosoft.com. Note that the path of the signaling messages may be completely different from that of the media exchanged between caller and callee. In this respect, SIP signaling differs radically from the typical telephone signaling, where the signaling message sets up state in each intermediate telephone switch [1]. Figure 1 typical SIP-based VOIP configuration and call flow III. SYSTEM MODEL AND WORK PROCEDURE The proposed network configuration used to study End-to-End delay, packet loss and jitter of SIP-based VOIP is as shown in figure 2. The effect of two different registrar servers (inbound and outbound SIP servers) is investigated over three different media (100 base TX, 10 base T, and wireless LAN 802.11g); we assume an ADSL connection to the internet with a speed 1024 kb/s to get an access to the outbound SIP server. Figure 2 Network configuration of a SIP Based VOIP Calls For the inbound SIP server a java based SIP server software is used (OnDO SIP Server SIP server with an "Educational Use" license by Brekeke Co) running on an INTEL Pentium 4 machine 2.8 GHZ processor and a memory of 512 MB DD Ram, The outbound SIP server is the WEB.de SIP server. The UAC is a soft phone (x-lite soft phone version 3.0 by COUNTER PATH ). Since the physical layer has an important effect in our study, we examined three different media types: wired Ethernet 100 base TX, 10 base T, and WLAN (wireless LAN) IEEE 802.11g for both registrar servers. The UAS is an analog phone connected via an ATA (analog terminal adaptor "fritz box fon ATA by AVM co") [4] to the Ethernet switch.a session or dialog is set up between two user agents following the SIP model shown in figure 1; where the requesting user agent client (UAC) interacts with the target user agent server (UAS). Figure 3 shows the flow of a typical SIP call (inbound registrar) where a UAC with an IP address 192.168.178.25 initiates a VOIP call by sending a SIP INVITE message to a UAS with an IP address 192.168.178.22. The INVITE message includes an SDP with the available audio codecs (G.711a law, G.711 u law, G.721 etc) that can be used in the session. the response to the INVITE message appears as Trying (100) and Ringing (180). Figure 3 also shows an RTP session between the two user agents which carry the audio traffic, SIP session is terminated at the end of the call using a BYE message and a response of Ok (200). SIP Request Messages and Response Codes are shown in Table 1, Table 2 [5].
an Ethernet LAN switch but via an access point with a speed of 54 Mbps (WLAN 802.11 g). The PC with soft phone initiates a VOIP call as UAC through the inbound or outbound registrar. Also the called person (UAS) is an analog phone connected to the network via an ATA,the packets from the PC NIC(wireless NIC) are captured and the packets delivered to the ATA is also captured using a network protocol analyzer. The captured packets is filtered to get the RTP packets flow in the media mentioned before; the previously mentioned three scenarios are repeated for both the inbound and outbound SIP server situations. Figure 3 Call Flow (Inbound Registrar Case) SIP Command Function INVITE Initiate Call ACK Confirm final response BYE Terminate and transfer call CANCEL Cancel searches and "ringing" OPTIONS Features support by other side REGISTER Register with location service Table 1 SIP Request Messages IV. EXPERMENTAL RESULTS Based on the above mentioned scenarios and the proposed SIP based VOIP network, tests were made from a client in a private network to clients in the same private network through a proxy or registrar in the same network (inbound registrar server). Response Code Prefix 1xx 2xx 3xx 4xx 5xx 6xx Function Searching, ringing, queuing Success Fowarding Client mistakes Server failures Busy, refuse, not available anywhere Table 2 SIP Response Code Three scenarios were carried out for each type of SIP server (inbound or outbound) over the three above mentioned media types Scenario 1 The PC with the soft phone connects to an Ethernet LAN switch with a speed of 100 Mbps (100 Base TX media). The PC with soft phone initiates a VOIP call as a UAC through the inbound or outbound registrar. The called person (UAS) is an analog phone connected to the network via an ATA (analog terminal adaptor) the packets from the PC NIC (network interface card) are captured and the packets delivered to the ATA are also captured using a network protocol analyzer (ethereal Version 0.99.0 is used as a network protocol analyzer from http://www.ethereal.com). The captured packets are filtered to get the RTP packets flow in the media mentioned before. Scenario 2 Similar to the previous scenario, the PC with the soft phone connects to an Ethernet LAN switch but with a speed of 10 Mbps (10 Base T media). The reason for this scenario is that 10 Base T media still exists up to now. It also puts more constrains on the delay and jitter due to is low bandwidth. Scenario 3 This scenario investigates the effect of using WLAN instead of wired LAN where The PC with the soft phone connects to Figure 4 transmitted and received voice signal in the WLAN scenario (outbound SIP registrar) In addition tests were carried out when the same client makes calls to clients in the same network through a SIP proxy and SIP registrar in a public network (outbound server). Results are collected using Ethereal (the packet analyzer software) for each call. Eleven different SIP calls were made and captured at both the source (UAC) and destination (UAS) interfaces for the different proposed scenarios using inbound or outbound registrar. From the packet analyzer software, RTP packets from both the forward (from NIC to the ATA) and reverse (from the ATA to the NIC) directions are filtered and converted into a SUN micro system audio file format (.au file)and so Both (forward and reverse directions) audio files are plotted as sown in figure 4. The figure shows the forward and reverse direction signals for WLAN case using outbound SIP registrar. The delay between the transmitted and received signals can be noticed and can be calculated using the cross correlation and autocorrelation. Also distortion and attenuation can be easily seen between the forward and reverse directions. Figure 5 shows the experimental result for the delay (for the different media types) encountered between the transmitted signal and the received signal for the mentioned eleven SIP
calls sorted in ascending order considering the delay. Figure 5 shows clearly that outbound registrar scenarios undergo higher delays than inbound scenarios for the three different media (100 Base TX, 10 Base T and WLAN). Delays for the eleven SIP calls using media (100 Base TX and 10 Base T) in inbound registrar case are nearly zeros and this is due to the high Bandwidth and absence of severe congestion in the inbound private network. Figure 6 autocorrelation and cross correlation between transmitted and received signals WLAN case (outbound registrar) Figure 5 Delay in SIP based calls over various media types Figure 5 also shows that the WLAN case has the highest delay among the three media independent of the registrar server location which either inbound or outbound, while the WLAN scenario with outbound registrar has the highest delay among all the six scenarios. This is because of the congested public network (INTERNET) plus the WLAN delay. Figure 6 shows cross-correlation between the transmitted and the received signals in the WLAN case (outbound registrar) as well as the autocorrelation of the transmitted signal in the same case. The two peaks of the autocorrelation and cross correlation happen at two different instances and the difference between them refers to the delay between the transmitted and the received signals. VOIP networks are very sensitive to packet losses which affect QoS. Packet loss refers to the packets of data that are dropped by the network to manage congestion. We may got packets loss due to the change of the used media and location of registrar server Table 3 shows the Average no of packets per call (average of eleven SIP based calls) which include SIP request, SIP response and RTP payload packets for each media used and registrar location and the percentage of the packet lost in that media due to server location. In VOIP networks voice is treated as if it were data, so voice packets will be dropped just as data packets are dropped under severe traffic loads and congestion. Lost data packets can be re-transmitted but this is not an acceptable solution for voice packets, while each voice packet can contains up to 40 ms or as many as 80 ms of speech information.packet loss, then, can significantly reduce QoS in VOIP networks. Even a 1% loss can significantly degrade the user experience with the ITU-T G.711 voice coder, which is considered the standard for toll quality [6]. Media Registrar Location Average no of packets/call Lost packets % 10 Base T Inbound 343 0 % 100 Base TX Inbound 333 0 % Wireless LAN Inbound 324 0.93 % 10 Base T Outbound 297 4.42 % 100 Base TX Outbound 319 4.85 % Wireless LAN Outbound 319 4.28 % Table 3 packet loss effect due to registrar location change It is clear that in the inbound network lost packets is almost zero for the different used Medias this is because of no congestion in the inbound network while in the outbound network is due to the uncontrolled congestion a packet loss exist with a value ranger from 4.2 % to 4.8 % from the total packets transmitted Media Registrar Location SIP response packets Resent packets 10 Base T Inbound 77 0 100 Base TX Inbound 77 0 Wireless LAN Inbound 77 0 10 Base T Outbound 94 3 100 Base TX Outbound 89 4 Wireless LAN Outbound 95 1 Table 4 packet resend due packet loss effect in SIP response Also Table 4 shows zero resend packets due packet loss in SIP response shown in Table 2 for inbound networks because of
zero packet loss in inbound networks but in outbound scenarios resented packets exits and this is because of the packets loss in the congested outbound network. The IP network can induce varying delays to the received packets that is known as jitter [7, 3]. To control Jitter, RTP and RTCP provide information, such as time stamps and interarrival jitter values that real-time communications applications can use to compensate for jitter during a session. An application s jitter buffers use the time stamps and the interarrival jitter values to make adjustments so that a smooth, even flow of packets is received. Figure 7 shows the maximum jitter, for eleven different VOIP calls, ordered in ascending order for 100 Base-Tx medium (100 Mbps) using either Inbound or Outbound SIP registrar.we can notice that maximum jitter values for the 100 Base-Tx medium using Inbound SIP registrar are almost the same as the values of those using outbound SIP registrar, in most of the calls. Figure 8 shows the Maximum jitter, for eleven different VOIP calls, ordered in ascending order for WLAN (54 Mbps) using either Inbound or Outbound SIP registrar.we can notice that the maximum jitter values for the WLAN using Inbound SIP registrar are almost the same as those using Outbound SIP registrar in most of the calls. In other words maximum jitter values for the 100 Base-Tx and WLAN are almost the same independent of the SIP registrar location. Thus the maximum jitter is affected by the media types which directly affect the bandwidth of the media (10 Mbps or 100 Mbps or 54 Mbps) of the LAN and/or WLAN and is not affected by the registrar location. Figure 7 maximum jitter for 100 Mbps media with inbound and outbound registrar The RTP receiver keeps a reserve of samples in order to absorb the network jitter, instead of playing out all the samples as soon as they arrive. This reserve is known as a jitter buffer. The bigger the jitter buffer, the more jitter it can absorb, but this also introduces bigger delay. If jitter buffer size is too small, then many late packets may be considered as lost and thus lowers the voice quality and lowers the overall QoS. Figure 9 maximum jitter for 100 Mbps and 10 Mbps Medias with inbound registrar Figure 9 shows the Maximum jitter, for eleven different VOIP calls, ordered in ascending order for LAN 100 Mbps (100 Base-Tx medium) and LAN10 Mbps (10 Base T medium) using Inbound SIP registrar.we can notice that packets in 10 Mbps LAN meets higher jitter than the 100 Mbps LAN, this is due to the limiter bandwidth for the 10 Base T medium (10 Mbps) compared to 100 Base-Tx (100 Mbps) medium. Figure 8 maximum jitter for WLAN 54 Mbps media with inbound and outbound registrar In our work we investigate the maximum jitter over the various media for inbound and outbound registrar cases as jitter is an important factor affecting QoS. V. MEAN OPINION SCORE The quality of transmitted speech is a subjective response of the listener. A common benchmark used to determine the quality of sound produced by specific CODECs is the mean opinion score (MOS). Compressor / decompressor (codec) systems and digital signal processing (DSP) are commonly used in voice communications, and can be configured to conserve bandwidth, but there is a trade-off between voice quality and bandwidth conservation. The best Codecs provide the most bandwidth conservation while producing the least degrading of voice quality. Bandwidth can be measured quantitatively, but voice quality requires human interpretation, although estimates of voice quality can be made by automatic test
systems. With MOS, a wide range of listeners judge the quality of a voice sample (corresponding to a particular CODEC) on a scale of 1 (bad) to 5 (excellent). Table 5 shows the different audio codec, their complexity, bandwidth and their MOS score [7]. Codec Bandwidth Complexity MOS score G.711 64 Kb/s Very low 4.5 G.726 16-24 -32 Kb/s low 4.1 (32 Kb/s) G.729a 8 Kb/s Low-medium 4 G.729 8 Kb/s Medium 4 G723.1 6.3-5.3 Kb/s high 3.6 Table 5 different voice coding algorithms MOS In our 3 different scenarios we use the ITU G711 u law codec and our MOS results are carried out by 31 listeners judgments to get results for the three different medias used in our experiments (100 Bast-TX, 10 Base-T and WLAN (air interface)) our MOS result is as shown in Table 6. 802.11g encountered the highest delays in both cases (inbound and outbound registrar). The limited bandwidth of the 10 Base T affected the jitter in this media compared with the 100 Base TX media..the congested outbound network induce more packet loss than the inbound network this is because of the uncontrolled traffic in the outbound network. The outbound registrar case always has the highest Jitter compared with the inbound registrar case whatever the media type was; this is due to the congested, highly traffic outbound network.listeners can not notice the change in speech quality neither by changing the location of the registrar nor by changing the media type this can be seen from our MOS (mean pinion score) results Our work was done for the audio codec G.711 u law 64 kb/s and as a future work the audio codec G.711 a law 64 kb/s and G.726 32 kb/s and GSM 13 kb/s will be investigated for jitter and delay. Also an ns-2 (network simulator) simulation will be done and the results for delay and jitter will be compared with our experimental results. Media type 100 Base TX Inbound results MOS score varience Outbound results MOS varience score 3.37 0.24 3.48 0.0185 10 Base T 3.49 0.12 3.56 0.0169 WLAN 802.11g 3.42 0.06 3.54 0.0211 Table 6 variance & MOS results for 3 different scenarios using G711 u law coding algorithms Comparing results for MOS in both Table 5 and Table 6 shows difference between the standard MOS values for the used codec (G711 u law) and the lisener MOS opinion for the same codec. MOS results in Table 6 shows that the voice quality slightly differed up on media used and the registrar location, the Varience between the avrege of the 31 listeners opinion for each one of the eleven SIP based VOIP calls indicate that the liseners were more confidant in their opinin and judjments in outboud senarios while they are less confident in their judjments in inbound sernarios. In Table 6, the media 100 Base TX give the lower MOS results for both registrar locations (inbound & outbound) while it also has the higher varience between the other medias(10 Base T & WLAN 802.11g) this indicat the reason for the lowest MOS score- for the better media in packet loss, end to end delay and jitterwhich is the liseners uncertinity.from our results We can notice easily that MOS results did not affected hard neither by the location of the registrar (inbound or outbound) nor by the media type(100 Base TX, 10 Base T and WLAN ). REFERENCES [1] Henning Schulzrinne, Jonathan Rosenberg, the session initiation protocol: internet-centric signaling" IEEE comm.- unications magazine,vol.38, October 2000. [2] J.Rosenberg, et al, SIP: Session Initiation Protocol, RFC3261, http://www.ietf.org/rfc/rfc3261.txt, June2002. [3] Samir Chatterjee,, Bengisu Tulu,, Tarun Abhichandani,and Haiqing Li, SIP-Based Enterprise Converged Networks for Voice/Video-Over-IP: Implementation and Evaluation of Components" IEEE journal on selected areas in comm.- unications, vol. 23, no. 10, october 2005. [4] AVM Co "Fritz!Box Fon ATA Manual" http://www.avm.de/en/service/manuals/fritzbox/ Manual_FRITZBox_Fon_ata.pdf, January 2007. [5] Alan B. Johnston, Understanding the Session Initiation Protocol" Second Edition Artech House 2004. [6] Intel white paper "Overcoming Barriers to High-Quality Voice over IP Deployments" 2003 [7] Cisco Technical Notes Document ID: 18902 Understanding Jitter in Packet Voice Networks." http://www.cisco.com/warp /public/788/voice-qos/jitter_packet_voice.html" [8] Vincent Barriac, Jean-Yves Le Saout, Catherine Lockwood, "Discussion on unified objective methodologies for the comparison of voice quality of narrowband and wideband scenarios"http://portal.etsi.org/stq/presentations2004/090- Mainz_FT_objective_methodologies.doc" January 2007. [9] The ITU G series recommendation " http://www.itu.int/rec/t- REC-G/en " January 2007. [10] G. Camarillo, R. Kantola, and H. Schulzrinne, Evaluation of transport protocols for the session initiation protocol, IEEE Network, vol. 17, no. 5, pp. 40--46, Oct. 2003. VI. CONCLUSION AND FUTURE WORK We presented in this paper a practical method for testing SIPbased VOIP over three different media. The wireless LAN