H. Schulzrinne, "Voice Communication Across the Internet: a Network Voice Terminal", Technical Report, Dept. of Computer Science, U. Massachusetts, Amherst MA, July'92  Home/Search   Document Details and Download   Summary   Related Articles   Check

....arrive in time to be played back so that the added play back delay can be effectively minimized. 1. Introduction A significant technical problem created by the integration of voice and data in a packet switched network is the recon struction of voice data at a receiver as a continuous stream [2, 5, 8, 9]. This is generally done by playing back the voice data after a delay offset from the departure time at the source of the voice data stream. This delay offset is called the play back delay, and it is an estimate for the upper bound on delay a packet may experience. If the network imposes a ....

....the sampling of the delay distribution) We compare the VirtualQueue technique to McanDer and StDev, upon which many commonly found techniques are based. These techniques use a play back delay calculation that is based on the estimation of the delay variance of the received packets (similar to [5, 8]) MeanDer and StDev assume that the play back delay should be K (a constant) times the mean deviation or standard deviation of the interarrival times of packets, respectively. The simulations use trace driven data acquired with an experimental voice quality audio conferencing system called ....

H. Schulzrinne, "Voice Communication Across the Internet: A Network Voice Terminal," The NEVOT Audio Conferencing Software, Univ. of Massachusetts, 1992.

....However, the longer packets can be delayed, the more resilient the receiver is to adverse network conditions. The contribution of this work is a description of the implementation of a bu#er playout algorithm within a VoIP system. The playout algorithm implemented is almost identical to NeVoT [Sch92]. Better schemes in terms of playout delay have been documented [RQS00] but the simplicity needed for our scheme moved us to make this choice. We should point out that Sicsophone is a working implementation and this the algorithmic complexity is an important factor. We motivate this approach with ....

....if a packet is too late. If the read pointer has already passed the point where a packet should be, and it has not been written, then we know that this packet is late. Insertion is simply a modulo operation and a pointer copy. Using a ring bu#er in this manner is identical to that described in [Sch92] where the authors motivate their choice of using a circular bu#er for performance reasons. In order not to replay old data from the bu#er when no packets are being sent we write empty samples that sound like audible background noise into the bu#er so that the listener is aware the connection is ....

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Henning Schulzrinne. Voice communication across the Internet: A network voice terminal. Technical Report TR 92-50, Dept. of Computer Science, University of Massachusetts, Amherst, Massachusetts, July 1992. 14

....such as in the Time Division Multiplexing (TDM) of PDH and SONET SDH. As a consequence, clock recovery and dejittering must be performed in the receiver of the IP packets. While the problem of dejittering is somehow similar to the case of packet audio and packet video applications (e.g. [1, 14]) the problem of clock recovery cannot be solved with conventional algorithms based on Phase Locked Loops (PLLs) Under normal conditions of jitter in the Internet, the duration of the convergence period become prohibitively long with PLLs (in the order of minutes or more [9] for good ....

....delivery instant of every individual data unit. The receiver buffer handles the variability of delay. The design of the buffer depends on several factors: the network delay, the packing delay, the input rate and the maximum packet size. The design of the receiver buffer is a well studied problem [1, 14] which is not tackled here. The maximum packing delay is provided through RTCP messages, while the characteristics of the network delay (e.g. average and maximum value) has to be provided by the reservation protocol (or estimated in the case of best effort) The source clock recovery method has ....

H. Schulzrinne, "Voice Communication Across the Internet: A Network Voice Terminal", Technical Report 92-50, Univ. of Massachusetts, July 1992.

....now interlinks nodes with native multicast capability and nodes running mrouted. The MBone served as a testbed for multicast applications in the early 90 s. Some of the earliest MBone applications include video conferencing tools (vic [50] nv [26] audio conferencing tools (vat [39] nevot [66], rat [34] shared white boards (wb [40] mb [76] and session directory tools (sdr [31] These applications provided a platform where many research issues in multicast multimedia applications were studied. In 1998, most of these tools were re implemented. Their common functions were ....

H. Schulzrinne. Voice communication across the Internet: a network voice terminal. Technical Report 92--50, University of Massachusetts at Amherst, Amherst, Massachusetts, July 1992.

.... best of our knowledge, in the literature the tools based on adaptive playout adjustment schemes either do not consider security services (see e.g. FreePhone [2] or simply enable encryption of data by using the well known DES block cipher [24] and a key prefixed by the two parties (see e.g. NeVot [26], and rat [13] In addition, some other software packages working at the application layer are proposed to offer a secure audio communication over the Internet (see e.g. Nautilus [11] PGPfone [29] and Speak Freely [28] but they do not include mechanized adaptive playout adjustment schemes. ....

H. Schulzrinne, "Voice Communication across the Internet: a Network Voice Terminal", Tech. Rep., University of Massachusetts, Amherst (MA), 1992

....multicast enabled subnets connected via IP tunnels, which started operations in 1992. The initial success of the MBone motivated further interest in the development of tools that make use of that infrastructure. Examples of audio and or video conferencing tools are INRIA s ivs, Schulzrinne s nevot [1], Xerox PARC s nv [2] UCB LBNL s vic and vat [3] and more. There were also tools that required a different type of multicast service: whereas audio and video data can tolerate some amount of packet loss without severely degrading the quality of the reproduced signal, applications which involve ....

....complexity of Huffman codes. A.3 Bit Stream Syntax To protect the compressed data stream against loss of data we use Reed Solomon codes of length 255, of which 225 are data symbols. The syntax of this data, as well as the way in which data is distributed into packets, is shown in Fig. 8. energy[1] energy[2] energy[n] length[1] length[2] length[n] GOP sequence number Packet size Parameters of RS Code Pos. in GOP n bitstreams payload Fig. 8. Bit stream syntax. The packetization step in Fig. 6 produces a stream of bytes as with a syntax shown in the left figure. This continuous ....

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H. Schulzrinne, "Voice Communication across the Internet: A Network Voice Terminal," Tech. Rep. TR 92-50, Univ. of Massachusetts---Amherst, 1992.

....(and video and whiteboard) segments of many technical conferences and workshops are now carried over the MBone multicast network [8, 22, 31] Smaller, more interactive, group meetings are also frequently conducted over the Internet using these multimedia tools. Packet audio tools such as NeVoT [54], vat [23] and RAT [17, 28] operate by periodically gathering audio samples generated at the sending host, packetizing them, and transmitting the resulting packet (via UDP unicast multicast) to the receiving site(s) For efficiency, a source s audio is typically divided into talkspurts (periods ....

....that talkspurt. A packet is generated every 20ms during a talkspurt, and hence a missing dot at a 20ms interval within a talkspurt indicates a lost packet within the network. The delay traces shown in Figure 1, as well as all other traces reported in this chapter, were collected using NeVoT [54], an audio conferencing tool that allows both point topoint or multicast connections. NeVoT has a tracing mechanism that can collect timestamps of packets sent and received, RTP sequence numbers [53] and vat virtual timestamps of packets. In our experiments and simulations we used vat virtual ....

Schulzrinne, Henning. Voice communication across the Internet: A Network Voice Terminal. Tech. rep., Dept. of ECE, Dept. of CS, University of Massachusetts, Amherst, MA 01003, July 1992.

....in the presence of handoffs in the following subsections. 5.3. Packet audio performance The main goal of our experiments was to evaluate the effects of handoffs on Internet audio applications. Examples of these applications are vat (visual audio tool) 13] and nevot (network voice terminal) [31], built on top of the User Datagram Protocol (UDP) 25] and commonly used over the Multicast Backbone (MBone) 18] We wrote a simple benchmark program, udpbench, that sends a stream of UDP packets from one host to another. It reports at the receiving host any lost, duplicated, or out of order ....

H. Schulzrinne, Voice communication across the Internet: a network voice terminal, Technical Report TR-92-50, Department of Computer Science, University of Massachusetts at Amherst (July 1992).

....der verbrauchten Bandbreite zu erzielen. Unsere Lsung kann eingesetzt werden, um die modernen frame basierten Coder zu untersttzen, die mit dem AP C Verfahren nicht arbeiten knnen. 1 Die Probleme der Paketverzgerung und ihrer Schwankung wurden mit dem Konzept des Playout Buffers in [Ramj94] [Schu92], Schu95] und [MoKT98] behandelt. 2 Das derzeitige AP C Verfahren korrigiert nur einzelne Paketverluste, die im Internet und im MBone dominant sind. 3 Die anderen Sprachblcken eines stimmhaften Signals sind durchaus wichtig fr die Sprachqualitt. Jedoch wird der Verlust von diesen Sprachblcken ....

....a small increase of bandwidth consumption. Our speech property based FEC scheme can be applied to support coders that operate on fixed size of speech frames and cannot be supported by AP C. 1 The problems of packet delay and delay variation are addressed by adaptive playout buffer in [Ramj94] [Schu92], Schu95] and [MoKT98] 2 The current AP C scheme assumes that isolated packet loss is predominant in the Internet and in the MBone. Thus, AP C s performance decreases with increasing packet burst loss. 3 The other voiced frames are also important to the speech quality. However, the loss of ....

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H. Schulzrinne. Voice Communication Across the Internet: A Network Voice Terminal. Research Report, Department of Electrical and Computer Engineering, University of Massachusetts at Amherst, July 1992. Development of a Loss-Resilient Internet Speech Transmission Method 68 Diploma Thesis Nguyen Tuong Long Le

....distributed databases, distributed simulations, concurrent processing systems, and cache coherence protocols. ffl Multimedia transport service: With recent advances in network technology, it is now possible to use multicast networks for high bandwidth multimedia applications such as audiocast [50, 90], videocast [39] group video conferencing [42, 99] distributed whiteboards [41] interactive TV, and other multimedia systems (e.g. tele medicine systems) In contrast to reliable multicast, multimedia applications are loosely synchronized groups that require real time 1 data delivery, but ....

H. Schulzrinne. Voice Communication Across the Internet: A Network Voice Terminal. Technical Report UM-CS-1992-050, University of Massachusetts, July 1992.

....of its data into network packets, and hence, can optimize for loss recovery through intelligent fragmentation and framing. About the same time that ALF emerged, we and others developed a number of tools to explore the problem of interactive audio and video transport across packet switched networks [43, 44, 45, 46, 47, 48]. After several iterations of protocols and experimentation with audio and several different video compression formats, it became clear that a one size fits all protocol was inadequate [49, 24] Instead, a framework based on ALF emerged where a thin base protocol defines the core mechanisms ....

Henning Schulzrinne, "Voice communication across the Internet: A network voice terminal," Technical Report TR 92-50, Dept. of Computer Science, University of Massachusetts, Amherst, Massachusetts, July 1992.

....no longer fit well to the exponential model, which may in turn affect the packet loss rate at the same utilization gain. We recorded several telephone conversations as digitized audio files. Next we applied to the audio files with G. 729 Annex B Voice Activity Detector (VAD) 13] and the NeVoT [19] Silence Detector (SD) The resulting spurt gap distributions to a large extent depend on the type of silence detectors and the volume level. In most cases, the spurt distributions is slightly more heavy tailed than exponential, whereas the gap distribution deviates strongly from an exponential ....

....an IP phone User local user eth eth voice (tcpdump) eth Fig. 1. Gateway based telephone recording setup III. SILENCE DETECTORS A. Introduction to G.729B and NeVoT SD We examine two silence detectors: G. 729 Annex B VAD (Voice Activity Detector) 13] and the NeVoT Silence Detector (SD) [19]. Both of them use the energy in a voice frame as a first estimate in silence detection. NeVoT is based on the ISI VT audio tool [18] The NeVoT SD uses a threshold that is dynamically updated but constrained with a min and max value. It uses a small hysterisis as well as a fixed but configurable ....

Henning Schulzrinne. Voice communication across the Internet: A network voice terminal. Technical Report TR 92-50, Dept. of Computer Science, University of Massachusetts, Amherst, Massachusetts, July 1992.

....only if either the receiver can adjust its playout rate or if there are gaps in the media stream that can be stretched and shortened by the receiver. The latter technique is generally used for packet speech, with the pauses between talkspurts adjusted to reflect changing delay jitter conditions [47, 48, 49, 50, 51]. In addition to these network induced delays, the application and media coding may add significant additional delays. Details are beyond the scope of this chapter (see [52] for additional material and references) but may dwarf network delays in many situations. These delays are caused by coding ....

H. Schulzrinne, "Voice communication across the Internet: A network voice terminal," Technical Report TR 92-50, Dept. of Computer Science, University of Massachusetts, Amherst, Massachusetts, July 1992. 40

....any Internet drafts directory, e.g. ds.internic.net,and gaia.cs.umass.edu: ffl the specification itself ffl a profile for audio and video ffl a general design document A University of Massachusetts technical report is also available. It describes basic issues and an earlier version of NEVOT [1]. 4 Basic Operation 1. If you want to adjust the play or recording volume or change the input and output ports, you need to run a separate application. tkaudio exists for all NEVOTsupported platforms, but you may also use the following: System application DEC HP usr audio examples acontrol ....

H. Schulzrinne, "Voice communication across the Internet: A network voice terminal," Technical Report TR 92-50, Dept. of Computer Science, University of Massachusetts, Amherst, Massachusetts, July 1992.

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H. Schulzrinne, "Voice Communication Across the Internet: a Network Voice Terminal", Technical Report, Dept. of Computer Science, U. Massachusetts, Amherst MA, July'92

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Henning Schulzrinne. Voice communication across the Internet: A network voice terminal. 6 Technical Report TR 92-50, Dept. of Computer Science, University of Massachusetts, Amherst, Massachusetts, July 1992.

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H. Schulzrinne. Voice communication across the Internet: A network voice terminal. Technical Report TR 92-50, Dept. of Computer Science, University of Massachusetts, Amherst, Massachusetts, July 1992. 7

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Schulzrinne, H.: Voice communication across the Internet: a network voice terminal. Technical report, University of Massachusetts, Amherst (1992)

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H. Schulzrinne. Voice Communication across the Internet: a Network Voice Terminal. Tech. Rep., University of Massachusetts, Amherst (MA), 1992. http://www.cs.columbia.edu/~hgs/rtp/nevot.html

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H. Schulzrinne, "Voice communication across the Internet: A network voice terminal," Dep. Comput. Sci., Univ. Massachusetts, Amherst, Tech. Rep. TR 92-50, July 1992.

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Henning Schulzrinne. Voice communication across the Internet: a network voice terminal. Technical Report 92--50, Department of Computer Science, University of Massachusetts, Amherst, MA, USA, 1992.

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H. Schulzrinne, \Voice communication across the Internet: a network voice terminal," Technical Report, Department of Computer Science, University of Massachusetts, Amherst, MA, July 1992.

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Schulzrinne, H.: Voice communication across the Internet: a network voice terminal. Technical report, University of Massachusetts, Amherst (1992)

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H. Schulzrinne, "Voice Communication Across The Internet: A Network Voice Terminal", Technical Report TR 92-50, Dept. of Computer Science, University of Massachusetts, Amherst, Massachusetts, July 1992.

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H. Schulzrinne, \Voice communication across the Internet: A network voice terminal," tech. rep., Dept. of Computer Science, University of Massachusetts, July 1992.

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