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Performance Evaluation of MPLS TE Signal Protocols with
Different Audio Codecs
1
Mahmoud M. Al-Quzwini, 2 Shaimaa A. Sharafali
1, 2 Computer Engineering Department, College of Engineering, Nahrain University, Baghdad, Iraq
ABSTRACT
This paper studies the performance of MPLS networks TE signal protocols with different voice codecs including PCM (64 Kbps), GSM FR (13 Kbps), G.723.1 (5.3 Kbps), G.726 (16 Kbps), G.728 (16 Kbps), G.729 (8 Kbps) and IS-641(7.4 Kbps). Simulation results show that the MPLS network with CR-LDP TE signal protocol outperforms the MPLS network with RSVP TE signal protocol in terms of the total amount of received voice packets, voice packet delay variation, voice jitter, and the number of maintained calls for all voice codecs. The results also show that G.723.1 codec type gives better results in terms of the number of maintained calls, but with least voice quality compared with other voice codecs.
Keywords:MPLS, Traffic Engineering, VoIP, CODECS, CR-LDP, RSVP.
1.
INTRODUCTION
The high increase in the number of internet users made services such as telephone and television to reach their customers via the internet and this has been forcing Internet Service Providers (ISPs) to improve their quality of service. With this increase as well as the advances made in real-time applications (voice and video), the traditional routers have the challenges of providing the required high bandwidth, fast routing as well as quality of service support. Due to the challenges of traditional routers to provide these requirements especially for voice and video, methods such as the use of Multiprotocol Label Switching (MPLS) and so on are now used [1]. MPLS is not designed to replace IP; it is designed to add a set of rules to IP so that traffic can be classified, marked, and policed. MPLS as a traffic-engineering tool has emerged as an elegant solution to meet the bandwidth management and service requirements for next generation Internet Protocol (IP) based backbone networks [2]. WAN bandwidth is probably the most expensive and important component of an enterprise network, Network administrators must know how to calculate the total bandwidth that is required for Voice traffic and how to reduce overall utilization, a description in detail for coder-decoders (Codecs), codec complexity and the bandwidth requirements for VoIP calls. A codec is a device or program capable of performing encoding and decoding on a signal or digital data stream. Many types of codecs are used to encode-decode or compress/decompress various types of data that would otherwise use a large amount of bandwidth on WAN links. Codecs are especially important on low-speed serial links where every bit of bandwidth is needed and utilized to ensure network reliability [3].
2. RELATED WORKS
Analyzing and optimizing voice traffic over data networks have been a major challenge to researchers and developers, many techniques have been proposed based on analyses from real word and simulated traffic.
Mahesh Kr. Porwal [4] made a comparative analysis of MPLS over Non-MPLS networks and showed that MPLS have a better performance over IP networks, through this paper a comparison study has been made on MPLS signaling protocols (CR-LDP, RSVP and RSVP-TE) with Traffic Engineering by explaining their functionality and classification. The Simulation of MPLS and Non-MPLS network is done; performance is compared by with consideration of the constraints such as packet loss, throughput and end-to-end delay on the network traffic.
Ravi Shankar Ramakrishnan et al.[5] analyzed three commonly used codecs using peer-to-peer network scenario. The paper presents OPNET simulator and they were considered only in Latency, Jitter and Packet loss. They were able to present from the results that G.711 is an ideal solution for PSTN networks with PCM scheme. G.723 is used for voice and video conferencing however provides lower voice quality. Music or tones such as DTMF cannot be transmitted reliably with G.723 codec. G.729 is mostly used in VoIP applications for its low bandwidth requirement that’s why this type is mostly common on the WAN connections and to transport voice calls between multisite branches.
Md. Arifur Rahman [6] calculated the minimum number of VoIP calls that can be created in an enterprise IP network. The paper presents OPNET simulator designing of the real-world network model. The model is designed with respect to the engineering factors needed to be reflected when implementing VoIP application in the IP network. Simulation is done based on IP network model to calculate the number of calls that can be conserved
Sarmad K. Ibrahimet al. [7] studied the performance of MPLS networks with TE signal protocols in relation with voice codecs. Simulation were performed and compared for a multisite network with PCM and GSM based VoIP. Simulation results show that the MPLS network with CR-LDP TE signal protocol outperforms
http://www.cisjournal.org the MPLS network with RSVP TE signal protocol in
terms of both the total amount of received voice packets and the number of maintained calls for both voice codecs. The main goal of our research is to study the performance of traffic engineering signal protocols (CR LDP and RSVP)with voice codecs for voice over MPLS network.
3. TRAFFIC ENGINEERING IN MPLS
NETWORKS
Traffic Engineering (TE) is a mechanism put in place to control the flow of traffic in networks and it provides the performance optimization of the network resources. The main characteristics of TE are fault-tolerance, optimum resource 448utilization and resource reservation [8]. The basic objective of the consideration of TE is to improve quality of service of some applications and use the available network resources efficiently. There are some important factors, which are needed for TE. These factors are; Path Selection, Traffic Management, Direction of Traffic along Computed Paths and Distribution of Topology Information. The LSPs in the MPLS network are established and the labels are distributed on each of the hops along the LSPs before packets could be forwarded. The LSPs can be established either by explicitly routed LSP or control driven LSP. Control driven LSPs can also be referred to as hop-by-hop LSP and are set by the use of LDP protocol. Explicitly routed LSPs can also be referred to as constraint based LSPS (CR-LSPs), which are specified in the setup message. At each hop, a label request is sent to the next hop along the LSP [9].There are basically two protocols used to set CR-LSPs in MPLS. These protocols are; Resource Reservation Protocol (RSVP) and Constraint based routed LDP (CR-LDP).
3.1 Constraint Based Routed LDP (CR-LDP)
CR-LDP is an extension of LDP to support constraint based routed LSPs. The term constraint implies that in a network and for each set of nodes there exists a set of constraint that must be satisfied for the link or links between two nodes to be chosen for an LSP [10]. CR- LDP is capable of establishing both strict and loose path setups with setup and holding priority, path Preemption, and path re-optimization [11]. CR-LDP and LDP protocols are hard state protocols that means the signaling message are sent only once, and don’t require periodic refreshing of information. In CR-LDP approach, UDP is used for peer discovery and TCP is used for session advertisement, notification and LDP messages. CR-LSPs in the CR-LDP based MPLS network are set by using Label Request message. The Label Request message is the signaling message which contains the information of the list of nodes that are along the constraint-based route. In the process of establishing the CR-LSP the Label Request message is sent along the constraint-based route towards the destination. If the route meet the requirements given by network operator or network administrator, all the nodes present in route distribute the labels by means of Label Mapping message.
3.2 Resource Reservation Protocol (RSVP-TE)
RSVP-TE is an extension of RSVP that utilizes the RSVP mechanisms to establish LSPs, distribute labels and perform other label-related duties that satisfies the requirements of TE [12]. The revised RSVP protocol has been proposed to support both strict and loose explicit routed LSPs (ERLSP). For the loose segment in the ER-LSP, the hop-by hop routing can be employed to determine where to send the PATH message [10]. RSVP is a soft state protocol. It uses Path and RSVPcommands to establish path. The CR-LSPs established by RSVP signaling protocol in MPLS network is described by the following steps:
a. The Ingress router in the MPLS network selects a LSP and sends the Path message to every LSR along that LSP, describing that this is the desired LSP used to establish as CR-LSP.
b. In this process the Path and RSVP messages are send periodically to refresh the state maintained in all LSRs along the CR-LSP [13].
c. The LSRs along the selected LSP reserve the resources and that information is send to Ingress router using the RSVP message.
4. VOIP CODECS
There are many codecs available for audio, video and text. We used in our evaluations some of G.7xx of ITU-T standards for audio compression and decompression. Table.1 shows number of compression schemes.
Table 1: Common Audio Codecs
Algorithm Bandwidth /Kbps Codecs types PCM 64 G.711 RPE LTP 13 GSM FR ACELP 5.3/6.4 G.723.1 ADPCM 16/24/32/40 G.726 LD-CELP 16 G.728 CS-ACELP 8 G.729A ACELP 7.4 IS-641
The popular voice codecs used in the telecommunication industry are G.711 which is widely used in the PSTN environment [14, 15].G.711 represents logarithmic pulse-code modulation (PCM) with 8 bits samples for signals of voice frequencies, sampled at the rate of 8000 samples/second, on 64 kbps channel. Using G.711 audio codec for VoIP will give the best voice quality; as it uses no compression and it is the same codec used by all Public Switched network and ISDN lines. It sounds just like using a regular phone or ISDN phone. But this codec takes more bandwidth then other codecs, up to 84 Kbps including all TCP/IP overhead. However, with increasing broadband bandwidth, this should not be a problem [16].
G.723.1 is the ITU-T standard that specifies the coded representation for speech in PSTN using Algebraic Code-Excited Linear Prediction (CELP) coding rates at
http://www.cisjournal.org 5.3Kbit/s and Multiples Maximum Likelihood
Quantization (MP-MLQ) at 6.3 Kbit/s. The 6.3 Kbit/s provides very good voice quality whereas the lower bit rate provides good quality with some more functionality [17].
G.726 is the Recommendation for speech coding at 40, 32, 24, and 16 Kbit/s (variable bit rates) using Adaptive Differential Pulse Code Modulation (ADPCM) transcoding technique [18].
G.278 is the ITU-T Recommendation for speech coding at 16 Kbit/s utilizing Low-Delay Code-Excited Linear Prediction Coding (LD-CELP) [19].
G.729a this annex provides the high level description of a reduced complexity version of the G.729 speech codec. This version is bit stream interoperable with the full version, i.e. a reduced complexity encoder may be used with a full implementation of the decoder, and vice versa [20]. Offers toll quality speech at a low bit rate of 8Kbps using CS-ACELP (Conjugate Structure – Algebraic Code Excited Linear Prediction). However, it is a rather "costly" codec in terms of CPU processing time; therefore some VoIP phones and adapters can only handle one G.729 call (channel) at a time. This codec provides robust performance but at the price of its complexity. This can cause calls to fail if the user attempts to use three-way calling, or place simultaneous calls on both lines of a two-line device, and G.729 is the only allowed codec [16].
Another standard used in our evaluations is ETSI GSM. The GSM system uses Linear Predictive Coding with Regular Pulse Excitation (LPC-RPE codec). It is a full rate speech codec and operates at 13 Kbits/sec. As a comparison, the old public telephone networks use speech coding with bit rate of 64 Kbit/s [16].
And IS-641 speech coding is based on the ACELP (Algebraic-code-excited linear prediction) [21]. The bit rate of the speech codec is 7.4 Kbit/s. The standard has been superseded by TIA/EIA-136-410.
5. SIMULATION
The simulation environment employed in this paper is based on OPNET 14.5 simulator which is extensive and powerful simulation software. Figures1 and 2 show two different MPLS networks, each one is simulated with CR-LDP and RSVP TE signal protocols. To simulate real network environments voice, video, HTTP, FTP, DB, Telnet and Email applications are used in each network. The VoIP traffic is sent from source (voice 1) to destination (voice 2), the video traffic is sent from source (video 1) to destination (video 2), DB and HTTP traffic is sent from source (DB, HTTP) to destination (DB, HTTP server), remote traffic is sent from source (remote) and FTP traffic is sent from source (FTP) to destination (FTP, remote server). The network in figure.1 consists of six routers and four switches and in
routers in each network are connected with DS3 cable with data rate of 44.736
Fig 1: first MPLS network topology
Fig 2: Second MPLS network topology
Mbit/s. The end nodes are connected to the network via switches. Links of each switch are 100BaseT. The voice workstations use different types of codecs, namely, PCM (64 kbps), GSM FR (13 kbps), G.723.1 (5.3 kbps), G.726 (16 kbps), G.728 (16 kbps), G.729a (8 kbps), and IS-641 (7.4 kbps) each type of codecs simulated individually and their results shown in figures 3 through 16, for the sent and received voice packet. Figures 17 through 30 show results of packet delay variation and end to end delay. Voice jitter and main opinion score results are depicted in figures 31 through 44.
6. PERFORMANCE METRICS
In our simulations, we use the following metrics to evaluate the performance of MPLS network.
6.1 Mean Opinion Score (MOS)
MOS provides a numerical measure of the quality of human speech in voice telecommunications, with value ranging from 1 to 5 where 1 is the worst quality and 5 is the best quality. In our simulation, we compute MOS through a non-linear mapping from R-factor as in [22]:
Where; R : the
effect of impairments that occur with the voice signal, : the impairments caused by different types of losses occurred due to codec's and network, and : represents the impairment caused by delay particularly mouth-to-ear
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6.2 Packet end-to-end delay (De2e)
The total voice packet delay; De2e represent in this formula:
(2) Where , , , and represent the network, encoding, decoding, compression and decompression delay, respectively [23].
6.3 Jitter
The jitter is defined as the signed maximum difference in one-way delay of the packets over a particular time interval.
Let t(i) and t’(i) be the time transmitted at the transmitter and the time received at the receiver, respectively. Jitter is then, calculated as in [24]:
(3) According to equation (3), the jitter value can be negative which means that the time difference between the packets at the destination is less than that at the source.
6.4 Packet delay variation (PDV)
Packet delay variation plays a crucial role in the network performance degradation and affects the user-perceptual quality. Higher packet delay variation results in congestion of the packets, which can results in the network overhead.
PDV is defined as the variance of the packet delay, which can be, calculated from the following Equation [24].
(4) Where is the average delay of the n selected packets.
7. RESULTS AND DISCUSSION
7.1 MOS
The MOS values shown in table.2 and figures 31 through 44 indicate that the voice calls with G.723.1 codec have the less quality than calls with other types of codecs.
7.2 Number of Maintained Calls
The voice delay can be divided into three contributing components which are described as follows [3, 25]:
• The delay introduced by the G.711 codec for encoding and packetization are 1 ms and 20 ms respectively. The delay at the sender considering above two delays along with compression is approximated to a fixed delay of 25 ms; • At the receiver the delay introduced is from
buffering, decompression, depacketization and playback delay. The total delay due to the above factors is approximated to a fixed delay of 45 ms.
• The overall network delay can be calculated from the above sender and receiver delays to be 80 ms approximately (150-25-45).
The 150 ms represents the maximum acceptable end-to-end delay so that the quality of the established VoIP call is acceptable [25].
In this paper the traffic drop time is used to calculate the number of maintained calls. In all simulations, the values of the packet end-to-end delay when the traffic drops were all less than the maximum acceptable end to end delay. This will show the difference among different codecs.
Then the number of maintained calls = (drop time – start
time) /2 (5)
The voice call start at 10 sec, and the drop time for each scenario is shown in table.5, and the number of calls maintained are shown in table.6. The results show that the number of maintained calls when using CR LDP TE signaling protocol is greater than those with using RSVP TE signaling protocol with all codecs.
7.3 Jitter
Table.3 and figures 31 through 44shows the results of maximum jitter values for different scenarios. Except the GSM FR and G.726 codecs in the first network, it is observed that CRLDP TE signal protocol has lower values than RSVP TE.
7.4 Packet Delay Variation (PDV)
Table.4 and figures 17 through 30show the results of packet delay variation for different scenarios. Except the GSM FR codec, it is observed that CRLDP TE signal protocol has lower values than RSVP TE.
Table 2: Summary statistics of MOS values experienced
Codecs Types Network number MOS value CR LDP RSVP PCM 1 3.595 3.595 2 3.674 3.674 GSM 1 3.467 3.467 2 3.548 3.548 G.723.1 1 2.559 2.559 2 2.546 2.546 G.726 1 3.595 3.595
http://www.cisjournal.org 2 3.674 3.674 G.728 1 3.562 3.562 2 3.675 3.675 G.729A 1 3.000 3.000 2 3.059 3.059 IS-641 1 3.468 3.468 2 3.550 3.550
Table 3: Summary of maximum jitter values experienced
Jitter value (sec) Max. Network number Codecs Types RSVP CR LDP 0.0000947 0.0000020 1 PCM 0.0002061 0.0000461 2 0.0016100 0.0017600 1 GSM FR 0.0019630 0.0002930 2 0.0041170 0.0002120 1 G.723.1 0.0026980 0.0000410 2 0.0001278 0.0001930 1 G.726 0.0003929 0.0000340 2 0.0002050 0.0001730 1 G.728 0.0003496 0.0000180 2 0.0003990 0.0000370 1 G.729A 0.0003462 0.0000230 2 0.0003830 0.0000050 1 IS-641 0.0011320 0.0001330 2
Table 4: summary of voice packet delay variations
Table 5: summary of traffic drop time
Table 6: summary of number of maintained calls
Table 7: summary of End-to-End delay values
Packet Delay Variations (sec) Network number Codecs types RSVP CR LDP 0.0000660 0.0000030 1 PCM 0.0000493 0.0000200 2 0.0000670 0.0004080 1 GSM FR 0.0002900 0.0005940 2 0.0004600 0.0000370 1 G.723.1 0.0002220 0.0000030 2 0.0018100 0.0001900 1 G.726 0.0001870 0.0000060 2 0.0009330 0.0001710 1 G.728 0.0005940 0.0000020 2 0.0003190 0.0000023 1 G.729A 0.0001700 0.0001100 2 0.0002300 0.0000200 1 IS-641 0.0001410 0.0000070 2 Number of calls/sec Network number Codecs types RSVP CR LDP 14 45 1 PCM 15 57 2 15 46 1 GSM FR 15 154 2 46 225 1 G.723.1 15 225 2 18 78 1 G.726 16 76 2 17 78 1 G.728 15 75 2 15 75 1 G.729A 15 78 2 10 150 1 IS-641 15 153 2 Codecs types Network number
End-to-End delay (sec) CRLDP RSVP PCM 1 0.079467 0.076834 2 0.078400 0.087500 GSM FR 1 0.126400 0.114500 2 0.142964 0.124584 G.723.1 1 0.148105 0.137048 2 0.111531 0.123505 G.726 1 0.111500 0.112300 2 0.070800 0.128800 G.728 1 0.113900 0.141138 2 0.070900 0.081100 G.729A 1 0.081200 0.098600 2 0.127800 0.106300 IS-641 1 0.109300 0.103700 2 0.104046 0.116862
Traffic drop time(sec) Network number Codecs types RSVP CR LDP 38 100 1 PCM 40 125 2 40 102 1 GSM FR 40 318 2 102 460 1 G.723.1 40 460 2 46 166 1 G.726 42 162 2 45 166 1 G.728 40 160 2 40 160 1 G.729A 40 166 2 30 310 1 IS-641 40 316 2
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Fig 3: 1st network PCM send and received voice traffic
Fig 4: 1st network G.723.1send and received voice traffic
Fig 5: 1st network G.726 send and received voice traffic
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Fig 7: 1st network G.729 send and received voice traffic
Fig 8: 1st network GSM FR send and received voice
traffic
Fig 9: 1st network IS-641 send and received voice traffic
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Fig 11: 2ndnetwork G.723.1 send and received voice
traffic
Fig 12: 2nd network G.726 send and received voice traffic
Fig 13: 2ndnetwork G.728 send and received voice traffic
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Fig 15: 2nd network GSM FR send and received voice
traffic
Fig 16: 2nd network IS-641 send and received voice
traffic
Fig 17: 1st network PCM PDV and E2E Delay
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Fig 19: 1st network G.726 PDV and E2E Delay
Fig 20: 1st network G.728 PDV and E2E Delay
Fig 21: 1st network G.729 PDV and E2E Delay
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Fig 23: 1st network IS-641 PDV and E2E Delay
Fig 24: 2nd network PCM PDV and E2E Delay
Fig 25: 2nd network G.723.1 PDV and E2E Delay
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Fig 27: 2nd network G.728 PDV and E2E Delay
Fig 28: 2ndnetwork G.729 PDV and E2E Delay
Fig 29: 2nd network GSM FR PDV and E2E Delay
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Fig 31: 1st network PCM jitter and MOS values
Fig 32: 1st network G.723.1 jitter and MOS values
Fig 33: 1st network G.726 jitter and MOS values
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Fig 35: 1st network PG.729 jitter and MOS values
Fig 36: 1st network GSM FR jitter and MOS values
Fig 37: 1st network IS-641 jitter and MOS values
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Fig 39: 2nd network G.723.1 jitter and MOS values
Fig 40: 2nd network G.726 jitter and MOS values
Fig 41: 2nd network G.728 jitter and MOS values
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Fig 43: 2nd network GSM FR jitter and MOS values
Fig 44: 2nd network IS-641 jitter and MOS values
8. CONCLUSIONS
In this paper, the performance of MPLS traffic engineering signaling protocols CRLDP and RSVP have been investigated with seven types of codecs PCM, GSM FR, G.723.1, G.726, G.728, G.729a, and IS-641. The performances of these codecs have been presented and compared.
Performance analysis focused on voice metrics included voice MOS values, voice end-to-end delay, voice jitter, voice packet delay variation, and voice sent/received packets. The number of calls is calculated and compared for each codec.
Results have showed that the MPLS network with CR-LDP TE signal protocol has a noticeable performance advantage compared to the MPLS network with RSVP TE signal protocol. It is five times more than RSVP in terms of number of maintained calls in the 1st network, this performance difference increases as the network becomes larger as it becomes seven times in the second network. CRLDP has 1.12% less end to end delay of that of RSVP 9 in the 1st network and by 8.09% in the 2nd network. For voice jitter, CRLDP is 34.34% of that of RSVP in the 1st network and 8.3% in the 2nd network. The voice packet delay variation of CRLDP is 21.4% of that of RSVP in the 1st network and 44.88% in the 2nd one.
The performance of the G.723.1 codec has the highest number of calls than other codecs but with poor voice quality. The IS-641 codec has higher number of calls 66.667% of that of G.723.1 codec with fair voice quality. Other codecs have less number of calls approximately 33.333% of that of G.723.1 codec but with fair voice quality. These results are for the respective codecs when applied with CRLDP protocol, however a similar performance difference between the codecs is obtained with RSVP protocols.
Increasing the number of paths in the 2ndnetwork increases the performance advantage of the CRLDP over the RSVP. This is due to the RSVP scalability issue when there are a large number of paths passing through a node due to the periodical refreshing of the state for each path.
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