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HD Voice – Definitive FAQ

FAQ

Definitive FAQ

Technical clarity for engineering teams. Regulatory assurance for compliance officers. Strategic insight for telecom leaders.

HD voice smartphone service by AT&T, Verizon, T-Mobile.

USTelco HD Voice is engineered around the AMR-WB codec (Adaptive Multi-Rate Wideband), delivering full-spectrum voice fidelity from 50Hz to 7,000Hz. This codec is critical for AI/NLP performance, real-time transcription accuracy, and perceptual clarity.

In addition, we support:

  • G.711u (standard fallback)
  • G.722 (for legacy HD voice environments)
  • Pass-through codec enforcement—we do not transcode unless explicitly configured

“To leverage full HD fidelity, your SBC or UC/CCaaS platform must negotiate AMR-WB without fallback to G.711. We’ll validate this during interop.”

USTelco maintains compatibility with virtually every standards-compliant SIP platform that supports NNI or private cross-connect. We’ve successfully deployed with:

Enterprise SBCs:

  • Cisco CUBE (ISR/ASR series)
  • Ribbon / Sonus SBC 5000/7000
  • Oracle/Acme Packet
  • Audiocodes, Dialogic, FreeSWITCH, Kamailio, OpenSIPS

Cloud Platforms / UC/CCaaS:

  • BroadSoft (Cisco BroadWorks/BroadCloud)
  • Genesys Cloud CX
  • Five9, NICE, 3CX
  • Twilio BYOC, Zoom Phone, RingCentral MVP (via peering)

Voice AI / Transcription Engines:

  • Deepgram, AssemblyAI, AWS Transcribe, Google Cloud STT, Nuance, Whisper

If your platform speaks SIP and can peer via private interconnect, we’ll enable HD Voice routing with zero-hop integrity.

You’ll need the following baseline architecture:
  • SIP-enabled SBC or SIP-aware platform
  • AMR-WB codec support (preferred)
  • Private interconnect via one of the following:
    • MegaPort VXC
    • Equinix Fabric
    • AWS Direct Connect
    • Dedicated fiber cross-connect (within common PoPs)
  • Static IPs with DNS FQDN (required for trunk auth)
  • CLI delivery and outbound attestation capability
We do not support:
  • Public SIP trunking over internet
  • NATed VoIP endpoints
  • Hosted PBX systems without NNI access
  • Non-attested CLI routes
“Once we receive your SBC info and POP location, our NOC will pre-map latency, codec negotiation, and failover configuration.”
HD Voice traffic is signed and verified end-to-end using STIR/SHAKEN. USTelco is an authorized STI Signing Provider and fully registered in the FCC Robocall Mitigation Database (RMD). Our infrastructure:
  • Applies Attestation A or B depending on origin authority
  • Signs every SIP INVITE with digital identity token
  • Preserves CLI with no stripping or rewriting mid-route
  • Supports traceback, audit, and CDR export for compliance validation
Fully aligned with:
  • TRACED Act (STIR/SHAKEN mandate)
  • ITG traceback protocol
  • RAY BAUM’S + Kari’s Law (E911 compliance)
  • TCPA, TSR, and 10DLC routing policies
“If you’re still routing through grey paths or LCR-based arbitrage, you’re not just risking quality—you’re risking federal noncompliance.”
USTelco HD Voice is delivered under a Carrier Service Agreement (CSA) or Enterprise Interconnect Agreement, based on volume and compliance profile. Included:
  • Dedicated HD SIP trunk provisioning
  • AMR-WB codec rights (where licensed)
  • Access to CLI attestation engine
  • 99.999% SLA for availability
  • Geo-redundant SBC failover routing
  • 24/7 NOC + compliance response team
  • CDR + STIR/SHAKEN traceable logs
Pricing Models:
  • Session-based (monthly per concurrent call path)
  • Usage-based (per minute, with short-duration filtering options)
  • Hybrid or custom models for large-volume, AI, or federal customers
“Every agreement includes regulatory compliance, CLI integrity, and SLA-backed infrastructure. There are no shortcuts to trust.”
 

Start by submitting your profile on our Request Access page. Include:

  • SBC or platform type
  • POP / fabric location (Equinix, MegaPort, AWS DC, etc.)
  • Expected daily call volume
  • CLI attestation readiness
  • Use case (AI, CCaaS, federal, fintech, etc.)

Once submitted:

  1. Our engineering team will validate codec readiness and interop
  2. You’ll receive a SIP config template + test dial plan
  3. CLI signing will be enabled upon successful routing validation
  4. Your trunk will be provisioned to production under encrypted SIP w/ STIR/SHAKEN header enforcement

 

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