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Opus and Session Initiation Protocol Security in Voice Over IP (VOIP)

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EJERS, European Journal of Engineering Research and Science Vol. 4, No. 12, December 2019 Opus and Session Initiation Protocol Security in Voice over IP (VOIP) Siddarth Kaul, and Anuj Jain Abstract—The purpose of the paper is to clearly elaborate III. MONO AND STEREO CODING FOR OPUS how Opus Codec can be used as Voice over IP or Unified Opus supports both Mono and stereo coding within a Communication as known Opus is an Audio Codec which is royalty free and most versatile format of Audio Codec the single stream the reference encoder tries to make the optimal Opus codec is used for Interactive Voice and Multimedia decision on number of audio channel based on bit rate application, Opus codec with WEBRTC (Web Real Time versus quality trade off as it always desirable to encode Communication) is a framework based on the Chrome Web input stream for enough quality the stereo decoder outputs Browser the codec behavior is usually effectively utilized under identical left and right channel upon decoding a stereo bit testing conditions for understanding the MOS assessment in stream [2]. comparison to the Opus Codec.The Opus codec is generally a low latency codec used for real time interactive communication the Opus codec replaces both Vorbis and Speex for new A. Packet Loss Resilience applications. Interframe correlation and the AMR-WB loss of opus Index Terms—Codec, CELT, Session Border Controller codec is around 30% the opus format is a combination of (SBC), Session Initiation Protocol (SIP), Unified full bandwidth CELT codec and the speech-oriented silk Communication, Cloud Communication. format both heavily modified: CELT is based on MDCT using CELP techniques the CELT is also a lossless recovery codec. Opus is one of the Voice Codec selected as I. INTRODUCTION mandatory to implement in Webrtc as a codec Opus can Opus combines the speech oriented linear predictive support narrowband and up to stereo full band while using coding SILK algorithm with CELT algorithm for maximum low bitrates with high resiliency [3]. efficiency in interactive communication, the Opus codec B. Testing OPUS in WebRTC framework also has a Hybrid mode which uses SILK and CERT for super wideband and full band audio Bandwidths in the Hybrid mode the frequency between the two cores is normally 8 KHZ. The SILK layer codes the low frequency up to 8 KHZ. Opus supports seamless switching between all its different operating modes. The Opus Codec is an inherent component of WebRTC capable browser the Opus codec supports the constant variable bit rate encoding from 6 kbit/s to 510 kbit/s for frame sizes from 2.5 ms to 60 ms the Opus codec can stream up to 255 audio channels. The Opus codec can handle wide range of audio application including VOIP, VideoConferencing etc. It can scale from low bit rate narrow band to high quality stereo music [1]. II. AUDIO BANDWIDTH AND BIT RATES The opus codec supports input and output of various audio bandwidths as defined in RFC 6716 the table below Fig. 1. Depicts the internal structure of Google Chrome browser for RTC shows the bit rates of Opus codec. communication. TABLE I: BIT RATES OPUS CODEC The RTC communication functions are as under: - Bit Rate Range (Kb/s) Configuration 1. Audio and Video Capturing 8-12 Narrowband Speech 2. Session Establishment and P2P communication 16-20 Wideband Speech 28-40 Full band Speech 3. Transport Function 48-64 Full Band Mono Music 4. Video and Voice engine codec session 64-128 Full Band Stereo Music generation capability. The Skype for Business also uses Opus Codec for audio on top of Opus codec in ORTC the Opus codec has Published on December 5, 2019 native support in skype for business media stack for all Authors are with Bhagawant University, India. DOI: http://dx.doi.org/10.24018/ejers.2019.4.12.1625 27 EJERS, European Journal of Engineering Research and Science Vol. 4, No. 12, December 2019 operations5. Fig.2 depicts the codec flow in Opus which involves an HTTPS connection of opus with skype or Skype for business audio and video service, the SFB uses the media relay service over HTTPS for video codec negotiation the Opus codec is also used in Polycom VVX Business Media Phones. Many session border controller manufacturers also are now supporting the Opus Codec. Fig. 3c. Cisco Call manger sequence for service parameter configuration. Fig. 2. Opus codec in Skype Opus Codec supports the Cisco Unified Communication Manager (CUCM) on version 11 there are also various CISCO devices support OPUS these are CISCO phones 7811/7822/7841/7861. The service parameters for Opus Fig. 3d. Cisco Call manger sequence for enabling Opus Volume Protcol. Codec need to be enabled on Cisco Unified Administration Page (CUCM) under system and service parameters [7]. IV. OPUS CODEC FOR ASTERISK The Opus Codec for Asterisk exposes a few configuration options that allow adjustments to be made on the encoder the following option is defined for custom types within codecs configuration file. The option includes error correction. [Opus Playback rate] Type=opus Max_playback_rate=8000; Limit the bandwidth on the encoder to narrow band Fec=no; Do not include in-band forward error correction data The option of a constant Bit Rate [Myopus, Playback rate] Fig .3a. Cisco Call manger sequence for enabling Opus codec Type=opus Bitrate=16000; Maximum encoded bit rate used V. OPUS CODEC IN LINEAR PREDICTIVE CODING Linear Predictive Coding involves the autoagressive modelling method the signal processing of Opus Codec is formed as a response to mostly IIR (Infinite Impulse Response Filters) the autocorrelation analysis. Figure 5 indicates the operation is divided so that the crossover sample rate is 16 KHZ in other words the input Fig .3b. Cisco Call manger sequence for enabling MOH. signal is divided into two coding paths in which it is decimated to 16KHZ sample rate for SILK and CELT frequency. The coding is divided in two branches represents the encoder is delayed in the D block to match the different times, Opus uses additional internal framing to allow the packing of multiple frames into single packet the MUX and DEMUX input output represents the Bit by Bit encoding of audio data stream in the OPUS Protocol. DOI: http://dx.doi.org/10.24018/ejers.2019.4.12.1625 28 EJERS, European Journal of Engineering Research and Science Vol. 4, No. 12, December 2019 a(1) r(1) = r(0) r(1) r(2) …………… r(p–1) (1) transport protocol provides necessary parameters for time sensitive exchange of metadata the protocol assigns a per a(2) r(2) = r(1) r(0) r(3) …………… r(p–2) (2) application playout buffer where packets are sorted for movingmedia stream ofeither a VOLTE or 4G or broadband a(3) r(3) = r(2) r(1) r(0) …………… r(p–3) (3) ready network. Opus seems to be more sensitive to jitter but performs better than Speex at extreme conditions, Opus has a(p) r(p) = r(p–1) r(p–2) r(p–3) …… r(n–1) (4) better voice quality at low jitter under heavier network perturbation [12]. Fig. 5. Block diagram of Opus Codec VI. SESSION BORDER CONTROLLER (SBC) OPUSTRANSCODING SUPPORT Opus audio codec as developed by IETF (Internet Engineering Task Force) which supports constant and variable bit rate support from 6 Kbit/sec to 510 Kbit/sec and Fig. 6. Correlation between Jitter and Subjective Quality of Experience sampling rate from 8 KHZ to 48 KHZ it incorporates The audio Jitter quality of experience in the above Figure technology from Skype speech-oriented SILK codec and 6 depicts that Opus codec used in 4G communication CELT codec. This feature adds the OPUS codec as well as networks will conceptualize design of VOLTE based support for translating and transcoding and pooled network the enhanced voice services comes out of 3GPP transcoding of different platforms in SBC [10]. working group the opus codec maximum packet width does A. Oracle Session Border Controller (SBC) Transcoding not fit into MTU without fragmenting into segmented Support packets into separate support of Opus Codec the maximum Opus is an Audio Codec developed by IETF that supports number of Bytes encoded in a bit stream is large as compare constant and variable bit rate encoding from 6 Kbit/sec to to capacity of MTU transfer the total signal to noise ratio 510 Kbit/sec and sampling rate of 8 KHZ to 48 KHZ. It needs to be precise and to the point so that sufficient bit incorporates technology from both skype speech-oriented stream can be transferred with message and audio packets to SILK codec and XIPH.The feature adds OPUS Codec as and fro [13]. well for support and Trans rating, transcoding11. SDP A. Testing of Opus Codec parameters rate specifies the sampling frequency. This Opus codec can be tested in two ways of functionality and parameter is mapped to the RTP clock rate in “a=rtpmap” performance the codec is implemented in a manner to be the range is limited to and must be 48000 Hz. tested through RAW PCM files that can be fed to API B. Sonus Session Border Controller (SBC) Transcoding functions and the performance was measured to evaluate Support how efficient the codec algorithms are and how much Opus audio codec is supported by various Sonus SBC in processors capacity is used for this purpose the different accordance with RFC 6716. Opus functionality is supported performance measurement parameters can be used as with an encoder and decoder format for various bandwidth constant value attributes for the purpose [13].
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