Voice DTMF provider moves services to the Cloud

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.
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• Bandwidth for both trunks supported collective aggregate for concurrency.
• TTBS SIP Trunk is a P2P 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 “To” header on the incoming calls.