Mobile VoIP Audio Quality CRASH COURSE THE INS AND OUTS OF MOBILE VOIP

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1 CRASH COURSE THE INS AND OUTS OF MOBILE VOIP

2 CRASH COURSE: TABLE OF CONTENTS THE INS AND OUTS OF MOBILE VoIP About VoIP...1 History of VoIP...1 Mobile VoIP WebRTC VoIP Sound Latency Jitter Call Setup Speed PSTN Routing VoIP Audio Additional Reading

3 1 CRASH COURSE: THE INS AND OUTS OF MOBILE VOIP Here at Sinch, we ve worked to make Mobile VoIP quality excellent for over 10 years. In this paper, I will give you some of the history, explain VoIP quality, and highlight the different areas that our experienced team is continually working on to improve VoIP connections. ABOUT VoIP A BRIEF HISTORY Voice over IP, or more commonly referred to as VoIP, is technology that enables voice calls using a broadband Internet connection instead of over a regular, phone line. VoIP can be connected to traditional local, long distance, mobile, or international numbers, to a traditional phone, or VoIP to VoIP over mobile and desktop applications. The Audio Transceiver, invented by VocalTec Communications, Inc., co-founders Alon Cohen and Lior Haramaty, is credited with enabling the creation of Voice Over Networks and ultimately, VoIP. As a result, VocalTec was the first to offer business phone services powered by VoIP back in 1995, forever disrupting telephony. Take a look at GetVOIP s nicely laid out history of VoIP and Internet Telephony

4 2 MOBILE VoIP Now that you know how VoIP got its start, let s skip ahead to 2003, when Swedish entrepreneurs Niklas Zennström, Janis Friis, and Estonians Ahti Heinla and Priit Kasesalu, launched Skype, peer-to-peer VoIP. Up until this point, VoIP telephony was enabled via Private Branch Exchanges (PBXs). Skype came onto the scene as the first consumer-level VoIP product, enabling people to make free calls to other Skype users over the Internet, and providing low rates to call phones and landlines, opening the door for other major VoIP applications from Rebtel to Viber to Tango. We believe that you should not have to pay for making phone calls in the future, just as you don t pay to send . CALLING DRIVEN BY VOIP NIKLAS ZENNSTRÖM Co-Founder of Skype

5 3 WebRTC WebRTC is the open-source standard toolkit for real-time media via Internet, which includes VoIP applications. To clear up any confusion between VoIP and WebRTC and how they interact, WebRTC makes it possible for web developers to enable VoIP into their applications. They are not competing products, but are often referred to in tandem because of their close ties. The technology behind WebRTC was developed by Global IP Solutions (GIPS), a company that delivered embedded solutions that enabled real-time communications capabilities for video and VoIP. The goal was to develop technology designed to mitigate the inherent challenges of VoIP from delay and jitter to packet loss, all of which directly affect audio quality in VoIP applications. Recognizing the technology s potential, Google purchased GIPS in May 2010 and released WebRTC as the open-source standard 11 months later. According to Smith s Point Analytics, WebRTC will be a $4.7B market by Producing high quality voice with stability to varying network conditions and ability to deploy on a broad range of devices such as mobile phones is a great challenge. Using webrtc software, Sinch tackles these challenges with great success Roar Hagen Co-Founder of GIPS, and integral Sinch team member

6 4 VoIP SOUND When we talk about quality, what are we talking about? At Sinch, we define quality by the performance of the following: CALL SETUP SPEED VOICE QUALITY LATENCY

7 5 VoIP SOUND LATENCY AND VOICE QUALITY An important factor when you re on a call is latency. High latency results in awkward pauses during the conversation, which often means you start to speak at the same time. The second factor is the quality of the sound. The less bandwidth you have, the less data you can transmit. That s where audio codecs come in. Audio codecs implement an algorithm capable of compressing and decompressing digital audio data and are included in WebRTC packages. CODECS Opus - This codec supports constant and variable bitrate encoding up to sampling rates of 20kHz bandwidth, which can reproduce the entire hearing range of the human auditory system isac A robust, bandwidth adaptive, wideband and super-wideband codec developed by GIPS OPTIMIZING THE OPUS CODEC We optimize by analyzing the quality of the audio and adjusting the max bandwidth usage in different steps. If audio quality is poor, we decrease bandwidth usage to slowly improve the sound and incrementally increase bandwidth as quality improves. We test on simulated degraded network conditions. This provides us with the ability to cope in subpar conditions and provide high voice quality with clear and crisp sound. We implement FEC (Forward error correction), which is a technique to reduce error by sending redundant data so that the receiver only receives the portion of data without errors. ilbc A narrowband voice codec also developed by GIPS VP8 Video compression technology developed by ON2 Technologies We use Opus, for its high-quality sound, variable encoding, and its low-data requirement. On its own, Opus is not automatically adaptive, but at Sinch we improve upon the codec with automatic and adaptive configuration to adjust latency and audio quality according to your network speed in order to optimize sound.

8 6 VoIP SOUND CALL SETUP SPEED SIP is standard protocol in telephony. At Sinch, we regularly integrate with partners over SIP or PSTN, but when we re talking about VoIP exclusively, we don t use standard SIP protocol because it s just not fast enough. With a standard SIP client, the call setup time is somewhere between 2-5 seconds. That means the caller will hear silence for 2-5 seconds before the phone even starts ringing on the other end. Instead, we use low latency call setup, which means the receiver is notified under a second when using our active connection technology and in 1-3 seconds using push notifications. JITTER Another factor to consider when discussing VoIP quality is jitter. Over VoIP, audio is transmitted in voice packets, and if packets arrive in the wrong order, callers can expect scrambled audio. Jitter is essentially the variability in latency (or packet arrival times). To combat scrambled audio, jitter buffers are required. These temporarily delay packets so that the end use can receive sound in the correct order with little distortion.

9 7 VoIP SOUND PSTN ROUTING When routing a call from VoIP to PSTN, or the public switched telephone network, which means you move from a call over Internet to a traditional phone call, Sinch monitors the system for early detection of connection issues or reduced quality and re-routes traffic accordingly. We also monitor Average Call Duration (ACD) and Answer Seizure Ratio (ASR). ACD: This is the measurement of the average length of calls, and when there is enough data can be telling of the connection quality as well as serve as a tool for forecasting traffic and call volume. Changes in ACD can be a strong indicator of changes in quality. ASR: This is the percentage of answered calls with respect to total call volume. A drop in ASR can indicate that a particular provider is having connection problems. When many providers are showing a drop to one country, this indicates in-country issues. The ability to isolate the reason behind a drop in ASR helps us determine if and when to switch providers. Given Sinch s large traffic volume and decade of historical data, we closely monitor ACD and ASR both per provider and destination in order to re-route traffic accordingly and continue to ensure our partners the best in quality.

10 8 OPTIMIZING VoIP AUDIO QUALITY The process of building great VoIP experiences on mobile is quite different from building browser-based solutions. The great thing about mobile devices is that they have unique hardware configurations, like two microphones on Sony Xperia or an excellent noise cancelation chip in the Samsung Galaxy. The variations of features built-in to mobile devices enable the Sinch team to take advantage when it comes to providing quality sound, but in parallel, these benefits contribute to the complexity of configuring VoIP calling. In order to tackle the numerous variables that can affect audio quality on mobile devices, we put in extensive work to optimize configuration. We test new handsets as they come out, configure each, and then automatically deploy the configurations without requiring code. With fewer variations in hardware, iphones pose less of a challenge, though we still test and automaticallyconfigure for any changes from Apple, as well. Audio quality is a top priority for us at Sinch. As VoIP is variable and quality is subject to your Internet connection, we focus on providing the best balance between latency and sound quality, while reducing call setup times. And when moving to PSTN, automatically routing traffic for increased optimization.

11 9 ADDITIONAL READING LEARN MORE ABOUT mvoip AND THE FUTURE OF VOICE CALLING WITH SINCH WEBRTC MOBILE FACEBOOK AND SLACK ARE IN 5 VOIP PROBLEMS THAT ARE NOW EXTINCT 18+ TOP WEBRTC NAMES TO FOLLOW ON TWITTER WHAT WILL THE FUTURE OF VOIP BE? HOW PEOPLE ACROSS THE WORLD ARE USING VOICE APPS CONTACT US Interested in adding VoIP calling to your app? Contact us to get started with Sinch. SAN FRANCISCO 180 Sansome St, 6th Floor San Francisco, CA U.S.A +1 (415) VOIP FACTS YOU NEVER KNEW

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