Abstract
This paper explores the opportunities of using not standardized compression codecs while transmitting the voice data over the existing network infrastructure of VoIP (Voice Over Internet Protocol) providers based on SIP (Session Initiation Protocol) in the Internet. Telecommunication networks offer a large set of services aimed at satisfying the ever increasing needs of users. One of them is a VoIP service implemented through network IP/TCP or IP/UDP stack. One of the most important determinants of this process is the need to use data compression. Codec implementation allows you to manage the data compression degree by changing the width of bandwidth usage and voice quality modeling. These features have priority when using VoIP on mobile devices on pre-paid cards such as smartphones where constraints are the size of the packet, the area signal strength in your area and the computing power of the device. Therefore, it is reasonable to develop new technology that provides flexibility in this regard. First of all, implementation of new codecs and ability to put them into VoIP calls according to SIP signaling standards influence on lowering the cost of deployment and then final costs of calls from subscribers located anywhere around the world with access to the Internet even at low bandwidth. The second advantage is the lower network load affecting the reduction of the costs of its maintenance by the owners of the network or service providers.
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ITU-T standards: G.711. Pulse code modulation (PCM) of voice frequencies, G.726. 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation (ADPCM), H.323. Packet-based multimedia communications systems
RFC3261: SIP: Session Initiation Protocol
RFC4566: SDP: Session Description Protocol
RFC3264: An Offer/Answer Model with Session Description Protocol (SDP)
RFC3550: RTP: A Transport Protocol for Real-Time Applications
Wielemborek, R., Laskowski, D., Lubkowski, P.: Effectiveness of Providing Data Confidentiality in Backbone Networks Based on Scalable and Dynamic Environment Technologies, Theory and Engineering of Complex Systems and Dependability, vol. 365, pp. 523–531. Springer, Switzerland (2015). ISSN 2194-5357
Lubkowski, P., Laskowski, D.: Selected Issues of Reliable Identification of Object in Transport Systems Using Video Monitoring Services, Communication in Computer and Information Science, vol. 471, pp 59–68. Springer, Switzerland (2014). ISSN 1865-0929
IETF Requests for Comments: RFC 3261. SIP: Session Initiation Protocol, RFC 6405. Session Initiation Protocol for Telephones (SIP-T), Voice over IP (VoIP) SIP Peering Use Cases
Lubkowski, P., Laskowski, D.: Test of the Multimedia Services Implementation in Information and Communication Networks, Advances in Intelligent Systems and Computing, vol. 286, pp 325–332. Springer, Switzerland (2014). ISSN 2194-5357
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© 2016 Springer International Publishing Switzerland
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Wielemborek, R., Sobieraj, T., Laskowski, D. (2016). Compression Codec Change Mechanisms During a VoIP Call. In: Zamojski, W., Mazurkiewicz, J., Sugier, J., Walkowiak, T., Kacprzyk, J. (eds) Dependability Engineering and Complex Systems. DepCoS-RELCOMEX 2016. Advances in Intelligent Systems and Computing, vol 470. Springer, Cham. https://doi.org/10.1007/978-3-319-39639-2_48
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DOI: https://doi.org/10.1007/978-3-319-39639-2_48
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Online ISBN: 978-3-319-39639-2
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