CARRIER GRADE VOICE OVER IP BY DANIEL COLLINS PDF

Overview Edit RTP is designed for end-to-end , real-time transfer of streaming media. The protocol provides facilities for jitter compensation and detection of packet loss and out-of-order delivery , which are common especially during UDP transmissions on an IP network. RTP allows data transfer to multiple destinations through IP multicast. Real-time multimedia streaming applications require timely delivery of information and often can tolerate some packet loss to achieve this goal. For example, loss of a packet in audio application may result in loss of a fraction of a second of audio data, which can be made unnoticeable with suitable error concealment algorithms.

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Design Next-Generation Wireless Networks Using the Latest Technologies Fully updated throughout to address current and emerging technologies, standards, and protocols, Wireless Networks, Third Edition, explains wireless system design, high-speed voice and data transmission, internetworking protocols, and 4G convergence.

Configure, manage, and optimize high-performance wireless networks with help from this thoroughly revised, practical guide.

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RTP digunakan bersama dengan protokol lain seperti H. Protokol memberikan kemudahan bagi pampasan ketaran dan pengesanan penerimaan data di luar aturan, yang merupakan perkara biasa semasa pancaran melalui rangkaian IP. RTP menyokong pemindahan data kepada pelbagai destinasi melalui multisiar. Sebagai contoh, kehilangan satu paket dalam applikasi bunyi mungkin menyebabkan kehilangan sebahagian sesaat data bunyi, yang boleh dijadikan tidak kelihatan dengan algoritma penyamar ralat yang bersesuaian.

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