VoIP Service Quality: Measuring and Evaluating Packet-Switched Voice
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Until the IP telephony industry matures to require apparatus and network performance guidelines and standards, the performance of IP telephony will be highly dependent upon the implementation of voice gateways and IP network performance. Therefore, meaningful and accurate measurement of voice GOS must take into account the performance impact of the IP telephony apparatus and the IP network.
An object of the invention is to provide a non-intrusive measuring method and apparatus to assess an end-to-end IP telephony network transmission quality. A further object of the invention is to provide a method for calibration of the apparatus and measurement algorithms associated therewith. Preferably, the method in step 1 further comprises using header information in the IP datagrams to smooth out any delay variation in the speech sample and computing a group of network performance parameters based on the IP datagram header information.
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In a preferred embodiment, step 2 further comprises determining a speech compression algorithm used in the IP datagrams by examining data encapsulated in the packets. The IP datagrams of the speech sample may be collected respectively at a far end IP interface point and a near end IP interface point. In accordance with another aspect of the invention, an apparatus for assessment of IP telephony networks transmission quality comprises:.
The apparatus preferably further comprises a data collector to collect the IP datagrams by identifying a packet flow associated with an IP telephony session. Preferably, the apparatus may be selectively co-located with a Personal Computer PC based IP voice gateway or stand-alone in the end-to-end IP telephony connection.
Also the apparatus preferably comprises a processor to correlate the voice GOS performance to IP network performance. In accordance with a third aspect of the invention, an initial calibration process for calibrating an IP telephony measurement apparatus comprising the steps of:. Preferably, the calibration IP datagrams are created from an IP telephony terminal during a calibration operation with a speech sample of known GOS and the output is a resultant GOS value.
This invention permits the IP telephony industry to measure a quality of the service they offer to their clients. Such a capability will provide not only a measure of voice GOS, but also a method of relating IP network performance to voice SLAs, which provides Internet Service Providers ISPs and Internet backbone network providers with guidelines for relating voice performance to network parameters that are both measurable and meaningful.
QoS (quality of service)
The invention exploits the connectionless nature of an IP telephony network in that the apparatus can be co-located in a voice gateway or deployed as a stand-alone apparatus, remote from a central location. The stand-alone apparatus would be required in an all-IP network scenario. Additionally, the invention takes advantages of the fact that most of the processes of the invention are already provided by voice gateways and any additional processes can be provided through software.
The invention will now be further explained by way of example only and with reference to the following drawings, in which:. The incident speech path and echo path are illustrated as well. That methodology is adapted to the field of IP telephony networks to provide a method and apparatus in accordance with the invention. The apparatus 32 are interfaced into an IP telephony network. In this embodiment, a far-end and a near-end public switched telephone network PSTN respectively interface into the IP network 26 through voice gateways The IP telephony measurement apparatus 32 are incorporated in each of the voice gateways.
At each end, the Signalling System 7 SS7 signalling network 36 also interfaces with the voice gateway 30 and PSTN 28 to permit detection of call progress states. Another embodiment of the IP telephony measurement apparatus 32 is a stand-alone configuration illustrated in FIG.
Voice over IP
An IP telephony terminal device 38 at each end interfaces directly with the Internet 34 via ISPs 40 instead of voice gateways The most common IP telephony enabled terminal devices 38 are personal computers with related IP telephony software and hardware. Consideration for a location of the IP telephony measurement apparatus 32 is the access to IP datagrams for a circuit to be monitored. It can be located at any point in the IP network where the IP datagrams are reliably collected. The IP telephony measurement apparatus 32 in the embodiments described above generally comprises the INMD 24 and software for pre-processing the IP datagrams collected from the IP telephony network and converting the datagrams into a format suitable for processing by the INMD A Digital Signal Processor may also be included to facilitate the processing.
The INMD in the IP telephony measurement apparatus 32 then processes the converted datagrams as it does in a connection oriented TDM telephony network to compute the GOS and other related network performance parameters. IP telephony measurement apparatus 32 performs a plurality of processing functions illustrated in FIG. The IP voice datagrams include packet header information such as source address, destination address and timestamp, and network performance information such as packet loss, delay and jitter can be derived using the packet header information.
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The protocol types of the IP voice datagrams are generally an H. A method for measuring voice GOS associated with the IP telephony measurement apparatus 32 comprises the following steps:. At the far-end IP interface point, the Internet protocol IP voice datagrams of a particular end-to-end IP telephony connection are collected by recognizing the flow and protocol types. As noted above, the most common protocol stack being an H.
At the far-end IP interface point, storing IP source and destination address information and a timestamp for post correlation analysis;. At the far-end IP interface point, using the RTP frame 44 header information including sequence number and timestamp to smooth out delay variation in the speech samples, which delay variation jitter is incurred in routing and switching apparatus during transfer through the IP network;.
At the far-end IP interface point, collecting parameters related to network performance in terms of delay, packet-loss and jitter relevant to a particular end-to-end IP telephony connection, the parameters computed being based on the sequence number, timestamp and length information included in the RTP header information;. At the far-end IP interface point determining a speech compression algorithm used to create the voice data, by taking the H.
Taking the smoothed-out compressed speech data and converting it to a format specified for INMD processing;. Computing, by INMD from the data prepared in steps 7 and 8, voice grade performance parameters including speech level, noise, echo and echo path delay, the far-end parameters being used to compute speech level and noise, while the near-end data being used to compute echo path delay and loss;. Co-processes may be involved in step 7 of the above process, depending on the IP telephony processes that need to be taken into account.
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The most common of the co-processes is error mitigation to remove the effects of error and packet loss. Another process is insertion of background noise during silent periods. Co-processes may also be involved in step 9, depending on the IP telephony connection. The most common of the co-processes in this step are active speech detection, tone detection, double-talk detection and echo cancellation.
It is both necessary and difficult to account for differences in implementations of IP telephony voice gateways and IP telephony enabled terminals, as related to the measurement accuracy of voice grade performance.
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Some of the main implementation differences relate to reducing path delay, poor network performance, mitigation techniques and echo cancellation. All these factors should be accounted for and are addressed by a method described below. The method is generally based on a performance factor that relates a difference in performance between the measurement apparatus being calibrated and a calibration standard. The measurement apparatus and algorithms for measuring voice analogue parameters require calibration.
The measurement apparatus 32 interfaces with the end-to-end known good voice network between two known good voice gateways 58 at the respective ends. The speech-source-to-gateway interface 60 , voice analogue interface 20 and two-to-four wire interface 62 all are known to be in good condition. The method of calibration involves measuring GOS ratings with the IP telephony measurement apparatus 32 using a voice IP network set-up known to produce good results.
Various tests exercise the complete range of speech quality parameters, such as speech level, echo path, noise, etc. Before calibration begins, an IP datagram calibration file is produced using a known good IP telephony terminal apparatus. The speech source 64 is the recommended speech samples specified by the ANSI standard. The IP datagram calibration file is used as an input to calibrate the IP telephony measurement apparatus The test cases are also specified by the ANSI standard. When the resultant GOS values match GOS values associated with the recommended speech samples within limits specified by the ANSI standard, the calibration is within acceptable limits of accuracy.
After the IP telephony measurement apparatus 32 is calibrated in the above known good IP network set-up, the IP telephony measurement apparatus 32 may be used, in conjunction with the IP datagram calibration file created during the initial phase of the calibration process described above to verify other voice gateways Persons skilled in the art will understand that the principle of the calibration is the same as the calibration of the IP telephony measurement apparatus 32 described above, even though the apparatus tested is a voice gateway The method for calibrating the IP telephony measurement apparatus 32 and algorithms for measuring voice analogue parameters comprises a process which includes the following steps:.
Creating an IP datagram calibration file including IP datagrams from an IP telephony terminal apparatus created during a calibration operation with speech samples of a known GOS;. This course should prepare you for starting a thesis project in this area for undergraduate students or beginning a thesis or dissertation for graduate students. This course will focus on the protocols associate with Voice over IP. The course should give both practical and more general knowledge concerning the these protocols. One of the major aims of the course is that student should be able to build upon these protocols to enable new services.
It is KTH policy that there is zero tolerance for cheating, plagiarism, etc. The course will mainly be based on the book: Henry Sinnreich and Alan B. Lecture notes are available on-line in PDF format. See the notes associated with each of the course topics. Errata for Henry Sinnreich and Alan B. Slides presented on IK vocabulary list - Last changed: Show versions. Add tag Aim This course will give both practical and general knowledge concerning Voice over IP. Enable you to utilize SIP in Presence and event-based communications Understand how SIP can provide application-level mobility along with other forms of mobility Understand how SIP can be used to facilitate communications access for users with disabilities for example using real-time text, text-to-speech, and speech-to-text and to know what the basic requirements are to provide such services Understand SIP can be used as part of Internet-based emergency services and to know what the basic requirements are to provide such services Contrast "peer-to-peer" voice over IP systems i.
Know the relevant standards and specifications - both of the protocols and of the requirements for example, concerning legal interception Understand the key issues regarding quality-of-service and security Evaluate existing voice over IP and other related services including presence, mobile presence, location-aware, context-aware, and other service Design and evaluate new SIP based services Read the current literature at the level of conference papers in this area. Demonstrate knowledge of this area both orally and in writing.
By writing a paper suitable for submission to conferences and journals in the area. Contents This course will focus on the protocols associate with Voice over IP. Topic selected Written report The length of the final report should be 10 pages roughly 5, words for each student; it should not be longer than 12 pages for each student - papers which are longer than 12 pages per student will be graded as "F". If there are multiple students in a project group, the report may be in the form of a collections of papers, with each paper suitable for submission to a conference or journal.
Contribution by each member of the group - must be clear in the case where the report is a collection of papers - the role of each member of the group can be explained in the overall introduction to the papers. The report should clearly describe: 1 what you have done; 2 who did what; if you have done some implementation and measurements you should describe the methods and tools used, along with the test or implementation results, and your analysis. Send email with URL link to maguire kth. Late assignments will not be accepted i.
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