Voice DTMF Provider Leverages SIP Trunk to Move Services to the Cloud
Highlights
- SIP Trunk: Business growth due to customer services being hosted on the private cloud, even during the lockdown
- Easily deployable solution due to IP technology
- Easily scalable solution with up to 1 500 channels
- Improvement in customer satisfaction
- Better voice clarity at 100 Kbps per channel
Introduction
The customer is an on premises DTMF voice solutions provider to BFSI clients. They build text to speech and speech to text solutions for their clients
Business Challenges
Due to the COVID 19 pandemic, the customer wanted to have their services hosted in a private cloud . The existing setup solution for their end customers was an on premise solution. Their text to speech and speech to text conversion solutions were hosted on the AWS public cloud.
The customer required a scalable and secure solution with better uptime and SLA
TTBS Solution
- TTBS delivered SIP trunk over Ethernet with SDH ring architecture
- SIP trunk terminated over SBC from the last mile
- Handoff Ethernet RJ 45 was connected to the SIP server (client server) with an inbuilt platform
- Both incoming and outgoing calls would take place with this hosted platform in a private cloud
- The path: Customer dials into a Direct Inward Dialling (DID) IVR plays customer responds through voice instead of dial pad voice is converted to text and sent to the master server (AWS) from where it receives a response and is sent back to the client server (SIP Server) converted from text to speech and the voice message response is finally delivered to the customer
- We provided 100 MB of ILL for seamless communication between the SIP server (client server) and the master server on AWS
- Bandwidth for both trunks supported collective aggregate for concurrency
- TTBS SIP Trunk is a P 2 P circuit with 30 IP address assigned The same IP address was assigned to the NIC interface of the customer PBX
- Codec on both trunks was configured along the internationally approved G 711 A law
- Dual tone multi frequency (DTMF) method followed RFC 2833.
- “P Preferred Identity” header was included in all SIP INVITEs for correct DID CLIP display on the outgoing calls
- DID no sending (DNIS) on SIP INVITE “ header on the incoming calls
- Keep alive mechanism was made available through SIP OPTIONS The expected response was SIP 200 OK from the CPE device
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TTBS provided the following apparatus
- Customer IP (configured at SBC’s NIC port)
- Gateway IP
- SIP Server IP
- Subnet Mask
- Username
- Password
- DID Range
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