Study and evaluation of voice over IP signaling protocols performances on MIPv6 protocol in mobile 802.11 network: SIP and H.323

2017 ◽  
Author(s):  
Azeddine Khiat ◽  
Mohamed El Khaili ◽  
Jamila Bakkoury ◽  
Ayoub Bahnasse
Keyword(s):  
Voice Over Ip ◽  
Signaling Protocols

Improved Jitter Buffer Management for WebRTC

2021 ◽  
Vol 17 (1) ◽  
pp. 1-20
Author(s):  
Yusuf Cinar ◽  
Peter Pocta ◽  
Desmond Chambers ◽  
Hugh Melvin
Keyword(s):  
Voice Quality ◽  
Buffer Management ◽  
Voice Over Ip ◽  
The Core ◽  
Management Algorithm ◽  
Core Concepts ◽  
Novel Strategy

This work studies the jitter buffer management algorithm for Voice over IP in WebRTC. In particular, it details the core concepts of WebRTC’s jitter buffer management. Furthermore, it investigates how jitter buffer management algorithm behaves under network conditions with packet bursts. It also proposes an approach, different from the default WebRTC algorithm, to avoid distortions that occur under such network conditions. Under packet bursts, when the packet buffer becomes full, the WebRTC jitter buffer algorithm may discard all the packets in the buffer to make room for incoming packets. The proposed approach offers a novel strategy to minimize the number of packets discarded in the presence of packet bursts. Therefore, voice quality as perceived by the user is improved. ITU-T Rec. P.863, which also confirms the improvement, is employed to objectively evaluate the listening quality.

Deconstructing Web Services

2012 ◽  
Vol 241-244 ◽  
pp. 3171-3174
Author(s):  
Chang Guang Shi
Keyword(s):  
Web Services ◽  
Virtual Machines ◽  
Voice Over Ip ◽  
Current Status ◽  
B Trees

Many experts would agree that, had it not been for telephony, the construction of B-trees might never have occurred. Given the current status of random theory, information theorists urgently desire the unfortunate unification of virtual machines and voice-over-IP, which embodies the unproven principles of robotics. We show that even though voice-over-IP and e-commerce can collaborate to achieve this goal, courseware and Internet QoS can synchronize to realize this mission.

scholarly journals Analisa Perbandingan Quality of Service Voice Over IP dengan Pengujian Codec Menggunakan Algoritma Low Latency Queuing

2017 ◽  
Vol 3 (2) ◽  
pp. 249-254
Author(s):  
Darmawan Darmawan ◽  
Yayan Syafriyatno
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Packet Switching ◽  
Voice Over Ip ◽  
Low Latency ◽  
Delay Jitter

Voice over IP (VoIP) adalah solusi komunikasi suara yang murah karena menggunakan jaringan IP dibanding penggunaan telephone analog yang banyak memakan biaya. Dalam penerapannya, VoIP mengalami permasalahan karena menggunakan teknologi packet switching yang mana penggunaannya bersamaan dengan paket data sehingga timbul delay, jitter, dan packet loss.  Pada penelitian ini, algoritma Low Latency Queuing (LLQ) diterapkan pada router cisco. Algoritma LLQ merupakan gabungan dari algoritma Priority Queuing (PQ) dan Class Based Weight Fair Queuing (CBWFQ) sehingga dapat memprioritaskan paket suara disamping paket data. Algoritma LLQ ini diujikan menggunakan codec GSM FR, G722, dan G711 A-law. Hasil pengujian didapatkan nilai parameter yang tidak jauh berbeda dan memenuhi standar ITU-T.G1010. Nilai delay rata - rata terendah yaitu ketika menggunakan codec G722 sebesar 20,019 ms tetapi G722 memiliki rata - rata jitter yang terbesar yaitu 0,986 ms.  Codec dengan jitter rata – rata terkecil adalah G711 A-law sebesar 0,838 ms. Packet loss untuk semua codec yang diujikan adalah 0%.  Throughput pada paket data terbesar saat menggunakan codec GSM FR yaitu 18,139 kbps. Codec yang direkomendasikan adalah G711 A-law karena lebih stabil dari segi jitter dan codec GSM FR cocok diimplementasikan pada jaringan yang memiliki bandwitdh kecil.

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