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![Auto attendants ( with rudimentary routing ) on freeswitch
<menu name="demo_ivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888@conference.freeswitch.org"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/>
<entry action="menu-exec-app" digits="3" param="transfer 1234*256 enum"/>
<entry action="menu-sub" digits="4" param="demo_ivr_submenu"/>
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/>
</menu>](https://image.slidesharecdn.com/currenttrendsandinnovationsinvoiceoverip3-200214101134/75/Current-trends-and-innovations-in-voice-over-IP-44-2048.jpg)
Uploaded byALTANAI BISHT
Current trends and innovations in voice over IP
The document discusses current trends and innovations in Voice over IP (VoIP) technology, detailing various tools and technologies such as SIP servers, FreeSWITCH, and machine learning applications in VoIP. It also covers the differences between VoIP and traditional telecom providers, the importance of call quality metrics, and methods for fraud detection in VoIP systems. Additionally, the document highlights auto attendants, the impact of over-the-top services on telecom providers, and debugging tools used in VoIP development.
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Similar to Current trends and innovations in voice over IP
Current trends and innovations in voice over IP
- 1. Current trends and innovationsin Voice over IP Webinar for OSFY Magazine 1 Feb 2020
- 2. Who am I? Altanai10 Years in Telecommunication Open Source Software enthusiast https://telecom.altanai.com/ tara181989@gmail.com Author of WebRTC Integrator’s Guide & Inventor of RamuDroid
- 3. Index 1. VoIP vsTelecom Providers a. SIP Servers types 2. Open-source tool and technologies in VOIP a. Opensip b. Kamailio c. Freeswitch d. Media Handling e. Webrtc 3. Machine learning in VoIP a. Call Classifier b. Fraud Detector c. NLP and Auto attendants 4. VoIP to telecom bridging
- 4. Voice over IP(VoIP) Methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.
- 5. VOIP vs Telecomproviders
- 7. SIP ( SessionInitiation Protocol) SIP Headers signaling and controlling multimedia communication sessions in applications of - Internet telephony for voice and video calls, - in private IP telephone systems, - in instant messaging over Internet Protocol (IP) networks and - mobile phone calling over LTE (VoLTE).
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- 9. SIP proxy servers Types: Statelessproxy servers Stateful proxy servers Forking proxy servers
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- 12. Kamailio SIP server https://www.kamailio.org/w/ SIPLoad Balancer, IP Telephony Engine, Least Cost Routing, SIP Firewall, Edge Proxy, SBC, Registrar and Location Service, Instant Messaging and Presence, MSRP, WebRTC, IPv4-IPV6, IMS, VoLTE
- 14. Opensips server https://www.kamailio.org/w/ SIP LoadBalancer, IP Telephony Engine, Least Cost Routing, SIP Firewall, Edge Proxy, SBC, Registrar and Location Service, Instant Messaging and Presence, MSRP, WebRTC, IPv4-IPV6, IMS, VoLTE
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- 20. Kamailio SIP serverswithout RTP proxy
- 21. Kamailio SIP serveralong with RTP proxy
- 22. RTP Proxy ( demoof freeswitch recording )
- 23. Codecs Codecs encode anddecode voice for network transmission – Algorithm (compression technology) – Bit rate – Sampling rate – Packetization 3 inbound negotiation algorithms – generous – greedy – Scrooge
- 24. Transcoding vs pass-thru Uses -Caller and callee is on diff codecs / bandwidths - Call recording and storing on file system - Tone detection - Play announcements
- 26. Freeswitch media manipulationmodules Mod_soundtouch - an adjust pitch , rate , tempo and other audio effects Mod_ladspa - use Linux Audio Developer's API inside freeswitch in realtime Mod_bert - provides bert_test, an application that generates and captures a milliwatt tone and prints an error rate at the end. Mod_portaudio - interfacing with sound card Mod_unimrcp - allows communication with Media Resource Control Protocol (MRCP) servers.media resources that can be controlled via MRCP: ● Automatic speech recognition (ASR) ● Text to speech (TTS) ● Speaker verification and identification (not currently supported by mod_unimrcp) Image from : https://freeswitch.org/confluence/display/FREESWITCH/mod_unimrcp
- 27. Transcription Mod_watson IBM’s Watsonmodule for freeswitch
- 28. Call Mixing -Conference bridges Mod_conference module on freeswitch
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- 30. WEBRTC WebRTC offers realtime communication natively from browsers. It is essentially just a media engine without control protocol Bodies behind standardization No plugins Supported browsers Supported Mobile platforms
- 32. Cloud communication Highlevel View
- 33. Machine learning inVoIP ● Call Classifier ● Fraud Detector ● NLP and Auto attendants
- 35. Implementation of MLfor VoIP Automatic gain control Echo cancellation Noise Suppression Call Classifications
- 36. Call Classification Classify callsbased on : Audio Quality metrics ● level ● gain ● noise ● harmonic and intermodulation distortion, ● frequency response, ● relative phase of signals ● interchannel crosstalk Call Quality metrics ● MOS score - packet loss , jitter , round trip time .. ● Energy Levels, ● Silence Thresholds, ● Silence Hits Applications ● Transcription and Sentiment analysis ● Automatic speech recognition ● Tone detection ● Speech scoring ● Speaker verification Classify Calls as ● Marketing calls ● Spam / Robo Calls ● Blank Calls ● Indoor / outdoor calls ● Emergency calls ● Monitor adult content / Lawful Interception
- 37. Call Fraud Detection ●Kamailio modules ● STIR/ SHAKEN
- 38. Kamailio modules forFraud and Spam detection Registration Hijacking HTTP authentication Impersonating a Server Proxy Authentication Temparing Message bodies end to end encryption over TLS + Digest Authorization S/MIME Denial of Service and Amplification Sanity Module - SIP syntax checking Pike Module - detects flooding Mid-session threats like tearing down session Full encryption vs hop by hop encryption ( show kamailio_conf.xml )
- 39. Robo / Spamcalls What is Call ID spoofing ? Vulnerability of existing interconnection phone system which is used by robo-callers to mask their identity or to make it appear the call is from a legitimate source, usually originates from voice-over-IP (VOIP) systems. STIR SHAKEN Standards developed by the Internet Engineering Task Force (IETF) and Alliance of Telecommunications Industry Solutions (ATIS) For telecommunication service providers implement certificate management system to create and manage the public and private keys, digital certificates used to sign and verify Caller ID details.
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- 43. Auto attendants Old WayIVR and DTMF Img source www.dialogic.com
- 44. Auto attendants (with rudimentary routing ) on freeswitch <menu name="demo_ivr" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu_short" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" confirm-macro="" confirm-key="" tts-engine="flite" tts-voice="rms" confirm-attempts="3" timeout="10000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="4"> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888@conference.freeswitch.org"/> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <entry action="menu-exec-app" digits="3" param="transfer 1234*256 enum"/> <entry action="menu-sub" digits="4" param="demo_ivr_submenu"/> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/> <entry action="menu-top" digits="9"/> </menu>
- 45. New NLU andNLP approach Image source :nextiva.com
- 46. NLP in forAuto attendants Applications : sentiment analysis Automatic essay assessment and scripting
- 47. VoIP to telecombridging
- 48. Features of OTTvs telecom providers Features by telecom providers ● Emergency dialing ● SMS / MMS ● Call ● Data ● VAS ● Billing ● DND
- 49. Telco Industry Trends OTP( over the providers ) are largely responsible to cut down profits of telco providers by providing free internet based communication platforms Source : .mckinsey.com
- 50. m Service Providersintegrating with voIP PSTN endpoints Gen band Telecom Providers Trunk Providers @alta nai VoIP Providers Internal PBX SIP endpo ints
- 51. 6. Debugging tools andQA 1. Chrome Dev tools 2. Selenium and Chrome driver 3. Wireshark 4. Tcpdump 5. VOIP clients 6. ffmpeg 7. test.webrtc.org -tbd- + Demo
- 52. References Telecom R&D https://telecom.altanai.com/ Freeswitch examples https://github.com/altanai/freeswitchexamples KamailioExamples https://github.com/altanai/kamailioexamples Opensips Examples https://github.com/altanai/opensipsexamples
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