Telnyx empowers businesses to combine HD SIP with a private infrastructure, enabling AI-powered calling across platforms, devices, and locations.
You call a customer service line and are greeted with audio that sounds like this:
Standard voice quality
Dissatisfied, you call that company’s competitor to ask about switching services and are greeted with audio that sounds like this:
HD voice quality
There’s no question which provider you’d rather do business with. Telnyx introduced high-definition voice to our SIP Trunking to help businesses meet the rising demand for customer service built on crystal-clear audio.
Most SIP providers still rely on narrowband codecs developed for legacy telephony networks. These codecs are reliable, but they compress voice signals into a range that is roughly half the spectrum of natural human speech, resulting in "telephone quality" audio.
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HD SIP expands the audio frequency range and employs modern compression algorithms, delivering voice quality that approaches in-person clarity. This comprehensive guide explores what HD SIP is, how it works, and why it matters for modern communications infrastructure.
HD SIP uses codecs like G.722 that capture and transmit a broader audio frequency, resulting in clearer, more natural-sounding conversations.
Narrowband codecs like G.711 sample audio at 8 kHz. This narrow band captures the essential speech frequencies needed for basic intelligibility but sacrifices the details that make voices sound natural and distinct.
HD SIP codecs double the sampling rate to 16 kHz, enabling clearer audio with better details. According to the ITU-T G.722 standard, wideband codecs provide improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders like G.711, which are optimized for POTS wireline quality of 300–3400 Hz.
| Feature | Standard SIP (G.711) | HD SIP (G.722) | Improvement |
|---|---|---|---|
| Sampling rate | 8 kHz | 16 kHz | 2× higher sampling |
| Frequency range | 300–3,400 Hz | 50–7,000 Hz | 2× wider bandwidth |
| Typical MOS score | 3.5–3.8 | 4.0–4.4 | +0.5 MOS improvement |
| Speech clarity | Basic intelligibility | Natural, in-person clarity | Reduced listener fatigue |
Unlike PSTN systems that were designed around circuit-switched networks with inherent bandwidth limitations, HD SIP operates natively on IP networks with flexible bandwidth allocation. This VoIP-native approach enables several key advantages:
Low-quality audio undermines user confidence in your capabilities before the conversation even begins. HD SIP delivers voice quality that feels contemporary and professional. The improved clarity reduces miscommunication and minimizes the need for repetition.
Research from TechTarget confirms that R-factor measurements, which convert to MOS scores, are especially useful for measuring improvements in voice quality when using wideband audio codecs. Today's VoIP management platforms calculate MOS and R-factor based on measured latency, jitter, and use of wideband audio.
HD audio allows agents to focus on problem-solving rather than interpretation. Low-friction communication correlates with improved agent performance and caller satisfaction, improvements that are measurable in key contact center metrics:
For organizations building AI-powered contact centers, explore how Telnyx contact center solutions deliver omnichannel, carrier-grade communications via API.
Voice authentication systems require consistent, high-quality audio samples to function properly. A study published in IET Biometrics found that wideband speech improves voice authentication precision by 1–3% of equal error rate over narrowband speech, even at the lowest investigated bitrates.
Accessibility technologies like hearing assistance perform significantly better with wideband audio input. And an expanded frequency range preserves accent nuances and pronunciation details, enabling robust support for multilingual customer bases.
As AI becomes central to contact center operations, HD SIP has become essential for meeting new technical demands. Real-time speech-to-text (STT) and natural language understanding (NLU) models depend on clean, high-frequency audio to perform accurately.
According to Deepgram's research on speech recognition accuracy, audio bandwidth directly shapes recognition quality. Narrowband audio (300 Hz–3.4 kHz) shows approximately 25% word error rate at 10 dB signal-to-noise ratio, while wideband audio achieves significantly better results—demonstrating a 13-point improvement that underscores how frequency capture shapes transcription accuracy.
Wideband codecs reduce the audio artifacts that interfere with transcription accuracy, which translates to measurably better AI performance across several dimensions:
For a deeper exploration of how audio quality impacts AI agents, read our guide on HD voice AI and why high-quality audio powers better agents.
Businesses whose operations demand clear audio and measurable performance gains should strongly consider upgrading to HD SIP in the near future. These use cases offer compelling arguments for HD SIP deployment.
| Use case | Key requirements | Expected outcomes |
|---|---|---|
| AI-first contact centers | Reliable STT, real-time sentiment analysis, AI-to-human handoffs | Lower word error rates, faster AI response times, improved agent coaching |
| SIP-based softphone deployments | Remote/hybrid workforce, software-based phones on desktop and mobile | Professional-grade audio quality leveraging better device hardware |
| Voice-enabled SaaS and IoT applications | Embedded voice functionality, developer flexibility | Scalable apps with professional audio quality |
| Cross-border PBX consolidation | Global telephony standardization, consistent quality across regions | Uniform call quality regardless of regional infrastructure |
AI-first contact centers need reliable STT performance, real-time sentiment analysis, and rapid AI-to-human handoffs. Improved audio quality also enables AI systems to provide better real-time coaching and conversation insights to human agents. Learn how to build conversational AI experiences with Telnyx's unified platform.
Remote and hybrid work environments rely on software-based phones running on both desktop and mobile devices. These endpoints typically have better audio hardware than traditional desk phones, making them ideal for HD SIP. For detailed setup instructions, see our audio and codecs configuration guide.
Apps that embed voice functionality require professional-grade audio. HD SIP gives developers the flexibility needed to build and scale these apps.
Organizations consolidating global private branch exchange (PBX) solutions face unique audio quality challenges. HD SIP helps overcome inconsistent regional telephony infrastructure, ensuring uniform call quality through an internet-based approach.
When evaluating HD SIP implementation, focus on technical metrics that track call quality and end-user experience:
Once you've identified if HD SIP is right for your business, the next step is finding a provider to help you build and scale your infrastructure. That's where Telnyx comes in.
Telnyx combines G.722 support with a private voice network engineered for consistent, low-latency performance.
The Telnyx Voice API automatically negotiates codecs between endpoints, prioritizing HD when supported and falling back to narrowband for legacy compatibility. Review our SIP trunking developer documentation for detailed implementation guidance.
Telnyx routes voice traffic over a global private network with strategically placed points of presence (PoPs), avoiding congested public internet paths. This reduces jitter and packet loss that would otherwise degrade HD audio quality.
Telnyx supports HD SIP across both traditional SIP trunking and modern programmable voice use cases, making it easy to deploy high-quality audio in mixed environments.
Explore the Telnyx Voice API for programmable voice applications, evaluate SIP trunking options for infrastructure modernization, or dive into the developer documentation to understand implementation details for your specific use case.
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