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VoIP Explained (3 hours)Forming part of mpirical’s acclaimed “Explained” series, this 3 hour eSeminar introduces VoIP. As such, this eSeminar will cover fundamental areas such as call establishment in VoIP scenarios, defining the use of protocols such as Session Initiation Protocol as part of this. In addition, the carriage of packetized voice using Real time Transport Protocol will also be detailed, as well as identifying a selection of key challenges faced by VoIP networks. Who Should AttendIT Staff requiring an understanding of how VoIP calls are established and how voice is transported. Technical Decision makers considering the challenges of VoIP deployment. SynopsisDivided into 3 sections, the topics covered will include: Section 1: VoIP Call Establishment (1 hour)This introductory section is designed to consider the establishment of voice calls using SIP (Session Initiation Protocol) and SDP (Session Description Protocol). As such, the key network architecture required for SIP is examined, focusing on elements such as the SIP Proxy or Call Manager. Additionally, SIP signalling flows will be employed to enhance understanding.
Section 2: Packetized Voice and RTP (1 hour)Following call establishment, this section is designed to show how the voice traffic itself is handled across the IP network. For this, RTP (Real time Transport Protocol) and RTCP (Real time Transport Control Protocol) are examined, showing how problems such as jitter, packet sequencing and QoS monitoring are supported.
Section 3: VoIP (1 hour)The concluding section contains a selection of VoIP deployment hurdles, explaining why tools such as NAT (Network Address Translation) causes problems for VoIP signalling. Also included is an overview of the challenges faced in terms of providing QoS and suitable security for VoIP traffic.
For more information: tel +44 (0) 1344 724666 or email enquiries@mpirical.com |
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