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

11  Download (0)

Full text


Vol. 4, No. 12, December 2019

Abstract—The purpose of the paper is to clearly elaborate how Opus Codec can be used as Voice over IP or Unified Communication as known Opus is an Audio Codec which is royalty free and most versatile format of Audio Codec the Opus codec is used for Interactive Voice and Multimedia application, Opus codec with WEBRTC (Web Real Time Communication) is a framework based on the Chrome Web Browser the codec behavior is usually effectively utilized under testing conditions for understanding the MOS assessment in 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 applications.

Index Terms—Codec, CELT, Session Border Controller (SBC), Session Initiation Protocol (SIP), Unified Communication, Cloud Communication.


Opus combines the speech oriented linear predictive coding SILK algorithm with CELT algorithm for maximum efficiency in interactive communication, the Opus codec 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].


The opus codec supports input and output of various audio bandwidths as defined in RFC 6716 the table below shows the bit rates of Opus codec.


Bit Rate Range (Kb/s) Configuration

8-12 Narrowband Speech

16-20 Wideband Speech

28-40 Full band Speech

48-64 Full Band Mono Music

64-128 Full Band Stereo Music

Published on December 5, 2019

Authors are with Bhagawant University, India.


Opus supports both Mono and stereo coding within a single stream the reference encoder tries to make the optimal decision on number of audio channel based on bit rate versus quality trade off as it always desirable to encode input stream for enough quality the stereo decoder outputs identical left and right channel upon decoding a stereo bit stream [2].

A. Packet Loss Resilience

Interframe correlation and the AMR-WB loss of opus codec is around 30% the opus format is a combination of full bandwidth CELT codec and the speech-oriented silk 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 mandatory to implement in Webrtc as a codec Opus can support narrowband and up to stereo full band while using low bitrates with high resiliency [3].

B. Testing OPUS in WebRTC framework

Fig. 1. Depicts the internal structure of Google Chrome browser for RTC communication.

The RTC communication functions are as under: - 1. Audio and Video Capturing

2. Session Establishment and P2P communication 3. Transport Function

4. Video and Voice engine codec session generation capability.

The Skype for Business also uses Opus Codec for audio on top of Opus codec in ORTC the Opus codec has native support in skype for business media stack for all

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

Siddarth Kaul, and Anuj Jain


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. 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 Codec need to be enabled on Cisco Unified Administration Page (CUCM) under system and service parameters [7].

Fig .3a. Cisco Call manger sequence for enabling Opus codec

Fig .3b. Cisco Call manger sequence for enabling MOH.

Fig. 3c. Cisco Call manger sequence for service parameter configuration.

Fig. 3d. Cisco Call manger sequence for enabling Opus Volume Protcol.


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]


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]


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 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.


Vol. 4, No. 12, December 2019

a(1) r(1) = r(0) r(1) r(2) ……… r(p–1) (1)

a(2) r(2) = r(1) r(0) r(3) ……… r(p–2) (2)

a(3) r(3) = r(2) r(1) r(0) ……… r(p–3) (3)

a(p) r(p) = r(p–1) r(p–2) r(p–3) …… r(n–1) (4)

Fig. 5. Block diagram of Opus Codec


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 sampling rate from 8 KHZ to 48 KHZ it incorporates technology from Skype speech-oriented SILK codec and CELT codec. This feature adds the OPUS codec as well as support for translating and transcoding and pooled transcoding of different platforms in SBC [10].

A. Oracle Session Border Controller (SBC) Transcoding Support

Opus is an Audio Codec developed by IETF that supports constant and variable bit rate encoding from 6 Kbit/sec to 510 Kbit/sec and sampling rate of 8 KHZ to 48 KHZ. It incorporates technology from both skype speech-oriented SILK codec and XIPH.The feature adds OPUS Codec as well for support and Trans rating, transcoding11. SDP parameters rate specifies the sampling frequency. This parameter is mapped to the RTP clock rate in “a=rtpmap”

the range is limited to and must be 48000 Hz.

B. Sonus Session Border Controller (SBC) Transcoding Support

Opus audio codec is supported by various Sonus SBC in accordance with RFC 6716. Opus functionality is supported with an encoder and decoder format for various bandwidth and call support for transcoding calls and for both support to 8, 12, 16 and 24 KHz. Opus.


Parameter Behavior

Maxaveragebitrate Mini (offer/answer of peer, route PSP,20Kbit/s) Useinbandfec In band FEC is used, if useinbandfec is set in the

route PSPand if the peer requests it

Used Tx DTX is used, if usedx is set in the route PSP and if the peer requests,it

Usedbr Constant bit rate if either peer requests cbr=1 OR route is configurated for cbr=1


Interactive real time audio streaming is very sensitive to timing parameters it’s very common to use specific protocols for media transmission. UDP based real time

transport protocol provides necessary parameters for time sensitive exchange of metadata the protocol assigns a per application playout buffer where packets are sorted for movingmedia stream ofeither a VOLTE or 4G or broadband ready network. Opus seems to be more sensitive to jitter but performs better than Speex at extreme conditions, Opus has better voice quality at low jitter under heavier network perturbation [12].

Fig. 6. Correlation between Jitter and Subjective Quality of Experience

The audio Jitter quality of experience in the above Figure 6 depicts that Opus codec used in 4G communication networks will conceptualize design of VOLTE based network the enhanced voice services comes out of 3GPP working group the opus codec maximum packet width does not fit into MTU without fragmenting into segmented packets into separate support of Opus Codec the maximum number of Bytes encoded in a bit stream is large as compare to capacity of MTU transfer the total signal to noise ratio needs to be precise and to the point so that sufficient bit stream can be transferred with message and audio packets to and fro [13].

A. Testing of Opus Codec

Opus codec can be tested in two ways of functionality and performance the codec is implemented in a manner to be tested through RAW PCM files that can be fed to API functions and the performance was measured to evaluate how efficient the codec algorithms are and how much processors capacity is used for this purpose the different performance measurement parameters can be used as constant value attributes for the purpose [13].

B. Functional Testing of Opus Codec

Opus codec functionality testing can be done with the help of module testing thus the codec module can be tested individually with a given input and output to a reference the module has an existing source code file as well as an input and output to a reference data binary files which can be used to input and output the data files for reference outputs.

Figure 7 shows a test bench procedure the test can be run in two phases in the first phase the input files are opened and loaded into memory after which they are encoded or decoded for the test cases the output files can be compared to reference files so that the output is identical otherwise the test fail. The PCM input format is a Opus codec which can be controlled parameter of the encoder the sample rate can be transmitted from test environment to configuration file


the test data normally used in the experiment should contain the opus packets the packet normally should flow in a transmitted test framework format only [14].

Fig.7.Module testing Block Diagram

The module tests were run with invalid input data these valid test results were used with internal data pointers the tests can be used for error handling situations the internal data framework as per the testing block diagram clearly indicates the valid use of error correction codes the error situation can be handled to avoid uncontrolled behavior [15].

C. Data Test Generation

Because there were no exact test vectors provided with the codec, they needed to be generated for essential aspect to cover all possible parameters the test can contain a PCM raw files with sampling rate of 48 KHZ furthermore it can encode speech samples of different varied types. In addition to continuous tests the decoder required data with silence or background noise periods. As a result, some of the test files are connected with several second gap test generation files.

The input data for the mobile networks is likely to contain the background noises the test cases were generated for audio to be tested should be encoded with reference application at different sample rates of 8 to 16 KHZ11. The bit rate selection should be of higher limits recommended in Opus standard for all audio bandwidths was set to a maximum of 510 Kbit/sec.

D. Performance Testing

The performance testing can be computed with the execution of speed for encoder and decoder the number of clock cycle elapsed for coding the real time processor analytics, the test application can be run natively on any LINUX platform and the application can be measured to the highest class of real time method used in affinity to the core set for DSP implementation15. The actual execution speed integrated to whole surrounding is the approximate time which can be used in DSP implementation the performance can be compared with the two platforms on which Opus can be tested this will help in DSP implementation of Opus the number of cycles does not indicate the execution speed totally because it is independent of the performance measured by computing how many cycles are consumed within a certain set of parameters when processing data the data is normally processed so that the different clock cycles are consumed with a certain parameter set when processing one second of data for multiple times[16].


The OGG OPUS is a new format of lossy audio codec format developed by Xiph foundation it was designed for a single stream of audio format with low latency the essential features include the metadata and the fast-accurate seeking, corruption detection after errors with the ability to multiplex Opus with minimal buffering. OGG bit streams are made up of streams of series of pages containing data from one or more packets each stream contains a checksum and capture packet for beginning and end of logical streams of different packets. The packet organization is organized into the first OGG Bit stream packets which uniquely identifies a stream as Opus codec on the first stream of the page. The second packet in the logical OGG bit stream containing header for supplied metadata the duration of the audio in the Meta data page should contain the opus packet in different streams.

The Granule position of the audio data has a total of 48 KHZ of audio sample rate the Opus encoder help minimize the jitter and other latency ratio [17].


The big data analytics for OPUS Codec can be used with various types of tools involving the analytical generation of different types of reports these reports can be generated through the various tools involving report analyzer in a segmented function involving CDR reports, Latency and network analysis records. Big Data analysis is used in Opus Codec is used in determent of the below depicted diagram which indicates the various parameters including Loss, Jitter and MOS values which smoothers the effect of packet loss to the acceptable result of 30 % loss. Opus codec involves performance better than 2%.

Fig. 8. Opus Depiction by Using Big Data Analysis

Opus Codec parameters running on communication services with Webrtc and other services. Opus codec hence is very different and relatively simple in terms of the different varying amount of traffic types and data types.

Opus codec in asterisk is also used in with new techniques for wideband and narrowband noise includes the maximum and minimum playback time or rate at which the end point allows opus for error correction at constant and variable playback rate [18].


Vol. 4, No. 12, December 2019



Max_playback_rate=8000; Limit the bandwidth on the encoder to narrow band

Fec=no; Do not include in-band forward error correction data.

X. OPUS CODEC APPLICATION IN IBMCLOUD Opus codec in the IBM Cloud is usually used involving the audio data is normally used with the sampling frequency of 16000 samples per second or equal to 16 Hz the Narrowband audio for real time application the service support to both applications involve the ideal analysis with help of Big data for real time information about Bit Rate flow in the cloud in Kbps and Mbps for two sampling rates of variable speeds. Big data analytics is also used in the compression and reduction of audio wavelength formats in lossless and lossy format of audio by as much as 10 times to the sampled rate of compression the audio / Ogg and Audio /Webm formats are the compression types which relies on the codec to encode Opus or Vorbis. The audio/wav format of compression can include uncompressed lossy or lossless data through various speech to text format [18].

Compression format types in IBM Cloud

Audio format and compression can be in three format Lossy, Lossless, None.

Content-type specification can be in required and optional format depending on the type of audio compression.

Required parameters can be None and Rate with Integer

Optional parameters can be None with Channel Endianness type Big and Little


Opus integration to Session Initiation Protocol can be performed with API provided by the developers of IETF it provides all the functionality needed for coding with Opus the Opus codec involves the API divided by its functionality and different interface levels in Encoder and Decoder communication and codec translation as depicted in the below Fig 9 [19].

Fig. 9 SIP and Opus API Integration

The API helps in Initialization, Processing and release of SIP data and codec translation which helps the processing of voice information in I/O. The Opus framework helps in error handling and information flow in different situations.

Opus codec will be able to transmit the encoded bit steam these frames are transmitted in encoded stream the packing can be performed within the codec with no external functionality needed the package normally consist of lower level of bit stream of UDP and RTP packet flow reducing the overall packet flow information of Opus packets. Opus

and RTP (Remote Transport Protocol) packing is used on top of UDP as IETF has standardized as how the RTP packet should be packed when the transmission of Opus in Bit stream, the time stamp increment of the RTP packet is normally set to the maximum possible limit the timestamp of the RTP packets has to be set to the maximum possible limit of around 48000 Hz per sample. Session Initiation Protocol Security threats with Opus Codec normally involve the client server model where the operations of the client and server involves the substitution transition of the packet flow between the CS and SC channel [20].


The CPN model carriers out tasks through different transactions including Invite and Non-Invite Transactions in our CPN model the top-level client substitution transaction will include two second level substitution Client Invite and Client _ NonInvite and the top-level server substitution transition will have second level of transaction for both the scenarios. In the model we use Client to Model the Invite to the server for the server Invite transactions the events including sending and receiving messages, timeout and error reporting are modelled with transitions the REQUESTS and RESPONSES model the channel from client to server. The CPN model for client request and responses is a substitution of the top to bottom approach of voice and video approach.

In the CPN model we use the client invite including sending and receiving messages through a substitution model the CPN model does combine the reachable state and occurrence sequences [21].

Fig. 10. Top Level CPN model with SIP


The tests done by Broadcom suggests the different configuration of CELT mode for Opus codec with MP3 and AAC_LC by using 10 diverse full band audio tracks with 44.1 KHz sampling used:

2 Pure Speech 2 Vocal

2 Solo Instruments 1 Rock and Roll 1 Pop

1 Classical Orchestra 1 Jazz


Fig 11 Opus Audio Test Results with SIP [21]


IMS uses Session initiation protocol to control the establishment of Voice and Multimedia sessions with SIP runs end to end as it originates and terminates end users’

terminals in the IMS the applications are implemented on a SIP server which may be augmented by logic on terminal equipment the SIP AS is a service execution platform on which one or more services are deployed it can be connected to control call function announcements [22].

Fig 12. IMS architecture [22]

A. Information regarding Call High jack

Call High Jack is performed by establishing a SIP based VOIP call between 2 SIP softphone users and an intruder using packet sniffer tools like Wireshark in absence of encryption. In the figure call high jacking is performed with the help ofusing the intercepting tool media and intercepting the flow of media packets both ways by the hijacker between the User A and the SIP registrar and Proxy server the call direction flow is normally Hijacked by the intruder to be changed to allow stealing of Information packets and data associated with flow of these packets. The registration of SIP packets along with Opus media is nowadays a part of asterisk and free source PBX and Skype also [23].

Fig.13. Opus and SIP Call Hijacking scenario

Opus and SIP (Session Initiation Protocol) normally do dissociate in different environments and both the protocol are tested in situations involving either the different tests either by flow type, data transfer and communication types.

The hijacking effects all scenarios in a condition where all the services involving development services, Operational services, Infrastructural and Application services are affected in a way that the message workflow and logging services are normally affected the storage, Clustering and Elasticity are also effected in a continuous code [24].

Fig. 14. SIP and OPUS service chart

Figure 14 describes the 6-step approach to the services of SIP with OPUS involving work load optimization, embedded management of services, in purpose programming model continuous agility involving rapid changes involve micro computation for health of these services.


The Opus codec is now days used in Microsoft teams.

The architecture of Microsoft teams involves legacy video, H.264 video tele presence, cloud phone, team’s client associated with a team combined ecosystem with MNP24 and SILK codec for teams’ conversation, teams messaging, calling and audio conferencing and video conferencing through SIP (Session initiation protocol and G711/G722 Codec support. The base of Microsoft teams involves Azure and share point services. The opus and SILK codec both involve the transparency data flow through which the background services of each and every protocol is evaluated on real time bases. The continues data stream of packet flow data information is communicated with Azure and share point cloud services.


Vol. 4, No. 12, December 2019

Fig.15a. Microsoft Teams Architecture with Opus and SIP

Fig. 15b. Teams type layer comparision

The sample rate for the different comparision for SILK wide protocol are as narrow band, medium band, wide band, super WB, full band with audio bit rates of 12-32 Kbps for mono and 24-64 Kbps for stereo.

Microsoft team’s communication flow is classified as in different sampling rates of 8 ,12 ,16,24 and 48 KHZ with different audio bands like narrowband, medium band, wideband, super-wideband and full band. Mono and Stereo encoder bit rates varies depending on their use with silk wide, Hybrid CELT. there are different encoding rates from 12kbps to 64 kbps.

The integration between service workload and application communication between the various flow levels for Microsoft teams to work involving share point, exchange, and skype in as a continuous work flow service. These all services are broken down into different workloads involving various file share options, regulatory compliance and unified communication features [25]. All these services independently of any technique work in continues momentum involving SIP as the primarily communication flow protocol with RTP media and opus protocol used for audio communication. The opus protocol continuously works with different other protocols and standard. The wideband integration of these protocols with SIP and opus protocol helps in better experimental results as depicted in the above diagram26. Opus and Session Initiation Protocol Security in Voice over IP (VOIP) clearly illustrates that the opus protocol is a loss-oriented audio coding format which clearly integrates with SIP standardized by ITF for low latency better speech analysis complexity and bit rate ratio.

The opus clearly improves the music and audio performance thus reducing the wideband audio gap to around 5m/s from over 100m/s being an open standard format the opus codec clearly has no software patents and is freely available under no license term which helps in easy integration of this codec format with rest of the media flow protocols. Opus has a high variety of usage by various organizations like

Microsoft, Broadcom, Huawei, and Qualcomm. All of these normally used the opus repository with SIP. The format of different modes involving speech, hybrid and CELT when using SILK audio frequencies for higher band width of above 8KHZ. SILK also supports with opus and SIP the different frame sizes of SILK from 10, 20, 40, 60m/s. frame shorter then 10m/s use CELT mode a typical opus packet can have transparency switch between modes, frame sizes, band width, peer packet analysis and differential cost and channel counts. The sampling and band width rate for opus helps in encoding and decoding the output sample rate for narrow band, medium band, wide band, super wide band.

Opus helped in standardization of different wide band and narrow band audio formats.25The development and standardization of opus audio coding format with SIP helped in development and standardization of various communication standards like skype, polycom video conferencing and cisco tele presence technology.

Representatives across the globe feels that a hybrid format always presents a new way of codec standardization in which submitted information either in the format of different communication mediums always helps in quality comparison and low latency performance of the new standard formulized for communication mediums across the globe. The direct routing for Teams is supported with a SIP for PSTN (Public Switched Telephone Network) which enables you to virtually any PSTN trunk and Microsoft Phone system and configuring the Interoperability between PBX (Private Branch Exchange) and Microsoft Phone system. A Hybrid Voice solution which involves the scheduled conferencing which requires proper licensing with online capabilities the direct routing supports the user’s phone system the domain name registered is your tenant it is possible for SIP address space in one tenant. Direct routing supports a wildcard with SAN which needs to conform to standard of HTTP over TLS.

The connection routing can be used with FQDN are as:

1. Sip.pstnhub.microsoft.com - Global FQDN 2. Sip2.pstnhub.microsoft.com – Secondary FQDN 3. Sip3.pstnhub.microsoft.com – Tertiary FQDN The FQDN are as above and normally are resolved using these IP addresses:







The firewall ports and protocols are as to connect a normal SIP trunk a possible proxy is used is 5061 you can use port number 5061 and a sip proxy for your sbc to port 1024-656536 with the media range of different addresses .The SBC (Session Border Controller) connects to SIP proxy which requires the Public IP where sip signaling is to be routed, Public DNS IP, Wildcard Support to DNS some of the SBC use TLS certificate with Public IP address and NAT is not supported with Direct Routing and Public IP address assigned to SBC[26].



The SIP media bypass in a direct routing scenario of where server is in the cloud is that the media is local, andthe media path is always optimal the media path is always supported by SBC providers with support to ICE light configuration at the SBC end and hear pinning with NAT is always supported for both directions. Media bypass externally is possible with SFB from outside the corporate network the client needs to be resolved with SBC on FQDN side the client can connect relay with SIP proxy. Media bypass in SIP with use of Proxy is considerably the most used phenomenon and nowadays is available with the server and the SBC it depends on the ways by which it will be configured for both the infrastructure usage methods to and fro [27].


A. Microsoft Teams and Opus Codec SIP media Bypass [28]

Example of a NS lookup of a SFB server involving Opus codec for Voice call joining through VNC remote desktop setup

Go to Run>NSlookup>Enter Server IP

Fig. 16. SFB server involving Opus codec for Voice call joining through VNC remote

B. Asterisk and Opus Codec SIP media Bypass Results


0=3666106261 3666106261 IN IP4 Sip media

C=IN IP4 B=A5/63

T=0 A=x nat 1

N×Audio 4009 RTP/AVP a*rtpmap 108 opus/4800

a*rtpmap 98 speex/1600

a*rtpmap 99 speex/32000 a*rtpmap 97 speex/8000

All credit for the original Asterisk patch to meet echoand forked by for Asterisk 11.20 or higher support.

The results as described above is the extension of SIP with Opus it allows the use of G.722 under Audio Codecs the test results have been conducted with use of a SIP IP phone where the phone always send an unauthenticated SIP invite and gets declined the Phone responds with 100 Trying and 180 ringing and 200 ok with cause code 16 this clearly shows that SIP with Opus and asterisk is only working in a particular scenario.29Understand that the Cloud Relay in and of itself really does nothing other than ‘phone home’ and wait for instructions. When it is first brought online and configured on the local network it will then immediately attempt to connect to a handful of hardcoded Fully Qualified Domain Names (FQDNs) which point to several services running across multiple Azure datacenters. If these connections are successfully established, then the new relay will then sit indefinitely in a holding pen, waiting to be manually integrated into a specific cloud tenant. Once this pairing step is completed by an administrator then the correct relay will be permanently linked to that tenant and begin pulling down any provisioned services which have already configured in the tenant. This includes the automatic download of any apps associated to the configuration, which are essentially docked into the Cloud Relay30. So in short, this relay is something that is simply brought online the first time using the local console and then from that point forward all management and configuration is performed through the appropriate cloud portal. Configuration changes and even software updates to the individual apps are all automatic. Currently the Cloud Relay itself is not updated so when new versions of the server image are released it would


Vol. 4, No. 12, December 2019 require the deployment of a new image, or replacement of

the existing. But the majority of the various service offering’s features and functionality comes from the individual apps which are automatically updated as stated.

The cloud configuration simply uses the capabilities to locate the source mailbox the authentication configuration mailbox creates credentials for each device to be well connected. The office 365 features involve the PowerShell cmdlets there are several commands which suffice the simple features of these services. Of the more recent changes which improve upon and simplify the overall management experiences there are two primary concepts worth calling out. One is the creation of a central repository for PowerShell resources and the other is the inclusion of Modern Authentication. The newer PowerShell Gallery is now used to store and distribute various modules making installation and updates of future module version much easier. Also, by leveraging Modern Authentication each of these modules utilize the same approach for providing administrative credentials for access31.

1) PowerShell Commands for SIP with opus implementation in Teams:

Get-CsOnlinePSTNGateway V1.0 tmp_v5fiulno.wxt New-CsOnlinePSTNGateway V1.0 tmp_v5fiulno.wxt Remove-CsOnlinePSTNGateway V1.0 tmp_v5iulno.wxt Set-CsOnlinePSTNGateway V1.0 tmp_v5fiulno.wxt Identity: sbc.contoso.com

Fqdn: sbc.contoso.com SipSignallingPort: 5067 FailoverTimeSeconds: 10 ForwardCallHistory: False ForwardPai: False SendSipOptions: True MaxConcurrentSessions: 100 Enabled: True

2) Command for Paired Gateway to appear with Options output

Get-CsOnlinePSTNGateway -Identity sbc.contoso.com Identity: sbc.contoso.com

Fqdn: sbc.contoso.com SipSignallingPort: 5067

Codec Priority: SILKWB, SILKNB, PCMU, PCMA Excluded Codecs:

FailoverTimeSeconds: 10 ForwardCallHistory : False ForwardPai: False SendSipOptions : True MaxConcurrentSessions : 100 Enabled: True

New-CsOnlineVoiceRoute -Identity “Redmond 1” - NumberPattern


(\d {7})$” -OnlinePstnGatewayList sbc1.contoso.biz, sbc2.contoso.biz - Priority 1 -OnlinePstnUsages “US and Canada”

Identity: Redmond 1 Priority: 1 Description:

NumberPattern: ^\+1(425|206) (\d{7})$

OnlinePstnUsages: {US and Canada}

OnlinePstnGatewayList: {sbc1.contoso.biz, sbc2.contoso. iz}

Name: Redmond 1 SuppressCallerId:


Set-CsUserContactList -Sip Address user1@domain.com -

ContactsSipAddress contact1@domain.com -GroupsToAddContactsTo

“Group 1” Adds contact1 to Group 1

Set-CsUserContactList -SipAddress user1@domain.com -

ContactsSipAddress contact2@domain.com, contact3@domain.com -

GroupsToAddContactsTo Group2, Group3 -AddToFavoritesGroupAdds contact2 and contact3 to Group2 and Group3 and Favorites

Set-CsUserContactList -SipAddress user1@domain.com -

ContactsSipAddress contact2@domain.com -GroupsToAddContactsTo Group3 -RemoveFromAllGroupsFirst removes contact2 from all groups then adds contact2 to Group3

Set-CsUserContactList -SipAddress user1@domain.com - ContactsSipAddress contact2@domain.com -

GroupsToRemoveContactsFrom Group3Removes contact2 from Group3

Presence <Presence Type>Specify the presence to set for the user. When setting the presence, it is invalid to set it to Unknown. -SipAddress<String>Specify the sip address of the user that you want to act on. The following formats are accepted: user@sipdomain.comsip:user@sipdomain.com- Server<String>Optional. Specifies the FQDN of the Skype for Business pool where the user is homed. Useful if automatic server discovery is not properly configured in your environment. One of the following DNS records needs to be configured in your internal environment to enable automatic server SRV.

3) User Presence Parameters

Get-CsUserPresence [-SipAddress] <String> [-Server <String>]


Set-CsUserTeamMembers -SipAddress user1@domain.com -AddMembers contact1@domain.com, sip: contact2@domain.com -

RemoveAllMembersFirst -DelayRingTime 10

First removes all contacts from the user’s team then adds contact1 and contact2 as team members. Sets the ring delay to 10 so that incoming calls wait 10 seconds before ringing the team.

Use this command to request and install the registration key for SEFAUtil Server. Below are the 2 ways to run this command:Set-SefautilServerRegistration -Name “contact name” -EmailAddress “contactemail@domain.com” - PhoneNumber “phone number” -ImplementationType Partner or PartnerImplement or SelfImplementThe above method requires Internet access to the following URL https://lcregistration.landiscomputer.comSet-

SefautilServerRegistration -RegistrationKey “paste key here”Contact sales@landiscomputer.com to request a key to use with the above method.To generate an auth key, the sip address of the trusted application endpoint is needed. To get that, run the following command: Get-Cs Trusted Application Endpoint -Application Id sefautil server [31].

C. Genesys opus SIP Interaction

If a Genesys call media server is used for interaction based recording the call media and recording uses administration to compress the recording of Opus the interaction recorder object is used for communication of G711 and G729 audio speech codec the media server completes the call recording operation and the system retrieves the call recording over HTTP and deletes the call recording from its location the workgroup is used for recording the different sets of calls as SIP protocol is pivotal in interaction of these recordings the Opus protocol helps in analysis of different audio wave ranges for these calls when the calls are recorded the Opus on a Genesys server enables the use of mono and dual channel recordings of high quality which utilizes the RTP and SRTP use of CIC servers which enables redundancy to one or more high interaction servers.

All file transfer including recordings use HTTP server


instead of file sharing the supervisory monitoring of calls works on automatic speech recognition of these remote media file types. Vorbis is also used with genesys SIP interaction protocol where sending and receiving the ASR media is possible with call flow protocol and media control mechanism for either T.38 and T.30 protocol for voice communication [32].

D. Packet Loss of Audio SIP system in Gensys

Packet Degradation – If a packet cannot read the information on RTP media the node discards the packet the degradation results in interference across the transmission media and problems of transmission. Long delays in single packets – Individual packets because long delays as in a VOIP system the discarded packets are replaced in an audio stream passed their system because their position in the audio stream is replaced with surrounding packets.

E. Windows event Logs

The windows event logs requests resources for interaction media to necessary create a call log used for a call with event details and event ID the probation is a condition where the interaction media analyses the event of logs.Interaction Media Server can enter the probation state for the following reasons: You configured Interaction Media Server to apply Quality of Service (QoS) to Real-time Transport Protocol (RTP) communications, but you did not install the Pure Connect QoS driver. The RtpPortRange property value on Interaction Media Server is too small and all available ports are currently in use. Use the Interaction Media Server Config-Properties page to configure the property. Another application is using User Datagram Protocol (UDP) ports and no other UDP ports are available for Interaction Media Server to service the interaction. If Interaction Media Server uses one network interface card (NIC) for Notifier traffic and one NIC for RTP traffic, Interaction Media Server cannot service the interaction if the RTP NIC fails.

Interaction Media Server has no available media engines to support the interaction. Interaction Media Server does not respond within a 10-second period to a resource creation request from CIC [32].


The various results show whether the Opus Protocol used with SIP in any of the technologies either Asterisk, Free PBX or Skype or Microsoft Teams and Gensys there is not much change in terms of understanding the security working needs of the either and the result also show communication itself within these technologies may change in terms of the audio bandwidth rates but the protocol working is not much effected we have observed through the test results that either the call hijacking is one of the scenarios where the SIP media flow is effected in either of the scenario and the RTP media packets do not have much effect in any scenario involving the media port opening and closing it is also clear that the various commands do not affect the call hijack scenario despite any change in media packet flow type we also observed that irrespective of the various communication flow mediums the Opus with SIP protocol have same response of media streams irrespective of different media

flow rates of 8000,10000 or 20000 KHz transfer data speech coding rates we also observed that signaling and media flow mediums are not impacted in even SIP trunks used in the communication mediums. WEBRTC service itself used with various communication mediums do not see much change in terms of protocol medium flow data types and rates of data medium change. The Big data analytics only help improve the analysis of the collection of data in and from various service providers thus providing a low cost high availability solution for better analysis of consequent solutions involving jitter, latency and packet loss information with high impact real time monitoring of these services in the event of resolution and quick information collection despite complex workflow and architectural reliability. Smart SBC and Protocol simulators and Protocol brokers actively help analyzing the tactics and media working reliability of these tools with ease of data collection and analysis, this difference is setup probes to monitor your network in real- time from SIP interfaces which decouples issues from vendor specific equipment and log files. The “box-level”

approach to monitoring better isolates issues regardless of vendors on your network or which SIP trunk provider you use. This also help define and resolve network equipment interoperability issues allowing you to focus on the service assurance. The idea of interoperability between those platforms and the Lync/Skype for Business platforms, both on-premises and online and teams continues to be a popular topic. While much has changed over time in terms of workflows and feature capabilities the overall need is no less important than before. In the event of the specific analysis and workflow there is not much need of change of all medium of different workflow scenarios These are individual on-premises server installations, some of which started as hardware appliances and were later also released as virtual servers, while others have been virtual servers since their inception. At this point all the components covered are available as software, where the MCU component could alternatively be deployed as hardware if desired in the event of clear instructions with the need by some or more specific needs as deployed.


[1] Doxygen Opusfile 0.7 https://opus-codec.org/docs/opusfile_api-0.7 [2] Flavio E Goncalves Building telephony system with Open SIPS pp 78-


[3] Salvatore Loreto and Simon Pietro Romano Real time communicationwith Webrtc chapter 5.

[4] Michael Maruschke, Oliver Jokisch, Martin Meszaros and Viktor IaroshenkoReview of the Opus Codec in a WebRTC Scenario for Audio and Speech Communication Conference Paper · September 2015

[5] Jenkins Refmanhttps://mf4.xiph.org/jenkins/view/opus/job/opus [6] https://www.microsoft.com/en-us/microsoft

365/blog/2015/09/18/enabling-seamless-communication-experiences- for-the-web-with-skype-skype-for-business-and-microsoft-edge/

[7] https://www.cisco.com/c/en/us/support/docs/unified- communications/unified-communications-manager- callmanager/introduction.

[8] https://www.cisco.com/c/en/us/support/docs/unified-


[9] Doxygen Opusfile1.2 https://opus-codec.org/docs/opus_api-1.2.

[10] J.Skoglund and M. GraczykIETF https://tools.ietf.org/pdf/rfc8486, October 2018.

[11] J.Spittika,K.Voctone,JM.ValinandMozillahttps://tools.ietf.org/pdf/rfc 7587, June 2015.


Vol. 4, No. 12, December 2019

[12] R. Bonica, C. Pignataro and J. Touch

https://tools.ietf.org/pdf/rfc7588, July 2015

[13] https://www.academia.edu/28092185/audio_and_speech_quality_surv ey_of_the_opus_codec_in_web_real-time_communication.

[14] H. Tscofenig, J. Arkko, D. Thaler and D. McPherson https://tools.ietf.org/pdf/rfc8486, March 2015.

[15] JM. Valin, k.Vos and

T.Terriberryhttps://tools.ietf.org/pdf/rfc6716,September 2012.

[16] Jean-


//jmvalin.ca/papers/aes135_opus_celt,2013 October 17-20.

[17] Koen Vos, Karstenvandborg,Sorensen, Soren Skak Jensen and Jean Marc Valin https://jmvalin.ca/papers/aes135_opus_silk. 2013 October 17-20.

[18] Eric Steven Raymond The art of Unix Programming Bell SystemTechnical Journal, v57 #6 part 2 (July-Aug. 1978) pp. 69-70.

[19] Sabu M thampi , Gregorio Martinez, and Carlos Becker West phallSecurity in Computing and communication 5th International Symposium pp 121-123

[20] Ilya GrigorikHigh performance Browser Networking pp 66-70 [21] Ted Wallingford Switching to VOIP O Reilly pp 55-58.

[22] Ian E Richardson VideoCodec Design Developing Image andVideCompression pp 66-69.

[23] Amruta Ambre and Narendra ShekokarDetection and prevention mechanism on call hijacking in voip system International Journal of Computer Applications 90(6) · February 2014

[24] Dan RisticLearning Webrtc pp 66-69.

[25] JeanMarcValin1,GregoryMaxwell1,TimothyB.Terriberry1,andKoenV os https://arxiv.org/pdf/1602.04845, 2013 October17-20,pp3-5.

[26] https://o365pp.blob.core.windows.net/media/Training%20Videos/Inte lligent%20Communications/Microsoft%20Teams%20%20Now%20A


[27] http://blog.schertz.name/author/jeff-schertz/page/1-2 on Microsoft teams

[28] Asterisk.forum.com/testresultscallestablishmentscenario.

[29] http://highsecurity.blogspot.com/2016/11/opus-vp8-codec-with- asterisk-1120-or.html.

[30] http://www.opusteam.co.uk/sip-trunks/callworkflowinformation.

[31] http://downloads.landiscomputer.com/sefautilserver/SEFAUtil- Server-Manual-en.pdf.

[32] https://help.genesys.com/cic/mergedProjects/wh_tr/desktop/pdfs/medi a_server_tr.pdf.

Siddarth Kaul born in Delhi 11/1/1988, received his Diploma in Electronics and Communication Engineering from Board of

Technical Education, Delhi, India, and Bachelor of Engineering degree in Electronics and Communication Engineering from GGSIP University, Delhi India in 2011, PGDCA from Sam Higginbottom Institute of Agriculture, Technology and Sciences, Allahabad, India MSC (IT) from Lovely Professional University, Punjab, India & MTech degree in Software Engineering from Singhania University, Rajasthan, India in 2013. He is presently pursuing Ph.D. in Computer Science Engineering from Bhagwat University, Ajmer, Rajasthan, India. He is a certified CCNA (Voice) professional and he has worked with TCIL (Telecommunication Consultants India Limited) on overseas project for two years as VSAT Engineer. He has also worked on Panterra Networks as Engineer Level 3 in Software Engineering Division. He had worked as Voice Engineer with Polycom, India before Joining Atkins (SNC LAVALIN Company). Presently he is working as Consultant Unified Communication in Accenture India. He is a unified communication SME in Voice and Video domain.

Dr Anuj Jain is presently working as Associate Professor in information technology in Lovely Professional University Punjab India. He is doctorate in computer science prior to his joining in Lovely Professional university Punjab India he worked with Bhagwant university Ajmer Rajasthan as assistant professor in Computer Science Engineering Department.




Related subjects :