Overcoming VoIP Quality Challenges Dr. Jan Linden, VP of Engineering Global IP Solutions

2. Overcoming VoIP Quality ChallengesDr. Jan Linden, VP of EngineeringGlobal IP Solutions

3. Outline • VoIP Quality Challenges • Latency • Codec Choice • Conferencing • How to Measure Speech Quality

4. VoIP Design Considerations Speech Quality Cost Quality Time to Market Cost Signaling Ease of Use Infrastructure Flexibility Network Impairments Features Device Considerations Power Consumption

5. Major Challenges for VoIP End-point Design Both Sides of the Call Need to be Considered Hardware Issues (Processor, OS, Acoustics, etc.) Speech Codec Codec Hardware Network Coping with Network Degredation Power Consumption VoIP Design Challenges Power Echo Echo Cancellation Environment – Background Noise, Room Acoustics, etc. Additional Voice Processing Components Voice Environment

6. Delay • Major effect is “stepping on each other’s talk” • Usage scenario affects annoyance factor – higher delay can be tolerated for mobile devices • Long delays make echo more annoying Impact of IP Networks • Packet Loss • Smooth concealment necessary • Network Jitter • Jitter buffer necessary to ensure continuous playout • Trade-off between delay and quality

7. Sources of Latency • Codec • Capture • Playout • Network delay • Jitter buffer • OS interaction • Transcoding Speech Encoding IP Interface Pre- Processing A/D IP Network Speech Decoding Jitter Butter Post- Processing D/A A/D Pre-processing Speech encoding IP interface IP Network D/A Post-processing Speech decoding Jitter buffer

8. Impact of Delay on Voice Quality • ITU-T (G.114) recommends: • Less than 150 ms one-way delay for most applications (up to 400 ms acceptable in special cases) • Users have got used to longer delays • Still, low delay very important for high quality Data from ITU-T G.114

9. Speech Codec Packet-loss Robustness Complexity Complexity • Many conflicting parameters affect choice of codec • Determines upper limit of quality • Support of several codecs necessary • Interoperability • Usage scenario • IPR issues a significant concern Memory Delay Speech Codec Input Signal Robutness Input Signal Robutness Bit-rate Bit-rate Sampling Rate Quality

10. Audio Spectrum • Better than PSTN quality is achievable in VoIP • Utilizing full 0 – 4 kHz band in narrowband • Wideband coding offers more natural and crispier voice Telephony band

11. Audio Spectrum vs. Speech Quality Speech Quality CDSpeech Super WidebandSpeech Wideband Speech Narrowband Speech (PSTN) Frequency 4 kHz 8 kHz 16 kHz 22.1 kHz 10 kHz

12. Speech Codec Design for VoIP • Many standard codecs designed for bit errors, not packet loss • Error propagation issue for CELP codecs • Variable bit rate attractive for IP networks • Packet overhead significant (5 – 32 kb/s) • Makes low bit rate codecs less attractive • Packet loss concealment a must • Jitter buffer design has significant impact on quality • Alternatives to standards • De-facto standards like iSAC • Open source like Speex

13. Echo Cancellation • High delay in VoIP makes echo problem more prominent • Network/Line echo cancellation for gateways • Acoustic echo cancellation • Hands-free/speakerphone • Small devices • Biggest challenge is AEC for PC • Acoustic setup unknown and changing • Wideband speech • Very few solutions on the market

14. Effects of Transcoding • Transcoding occurs when the endpoints are using different codecs • Every transcoding introduces distortion • Low bit-rate codecs very sensitive to transcoding • Transcoding between networks • Transcoding in conferencing • Mixing done in decoded domain results in transcoding VoIP to PSTN VoIP to Cellular VoIP to VoIP • Limited quality degradation since G.711 used on the PSTN side • Severe quality degradation common since low bit-rate codecs typically used on both sides • Usually occurs in Session Border Controllers • Can normally be avoided

15. How to Make the VoIP Software Robust? Very Quick Jitter Buffer Adaptation – Conditions Change Very Rapidly (on a milisecond basis) Spot Jitter Patterns - Increase Delay to Keep Good Quality when Unavoidable Quick Jitter Buffer Spot Jitter Pattens Minimize Delay Packet Loss Concealment Packet Loss Concealment - Capable of Handling Several Lost Packets in a Row Minimize Delay Everywhere – every milisecond counts

16. Measuring Voice Quality Subjective Methods Objective Methods • Test the “right thing”, i.e. subjective quality • Takes all types of degradation into account • Time consuming and costly • Lack of repeatability • Simple and affordable • Inaccurate but repeatable results • Sensitive to any processing (non-linear filtering, echo cancellation, time warping etc.) • Time synchronization major challenge not yet solved • Sensitive to background and equipment impairments • One step behind development of codecs and error concealment • Next generation algorithm in standardization process (P.OLQA)

17. Audio Conferencing • Design includes a trade-off between quality and scalability • Client based or server based • Server based offers better scalability than client based • Can be combined • Transcoding often unavoidable • Two strategies: • Mix incoming signals to form one output signal • Only relay packets and mix at client side • Multi-codec support • In relay mode all endpoints need to support all codecs • Narrowband and wideband • Both can be present in a conference • Narrowband participant will hear everything in narrowband • Wideband participant hears others in narrowband or wideband E A+B+C+D A A+B+C+E B+C+D+E D A+B+D+E A+C+D+E B C

18. Conclusions • Latency has a significant impact on the perceived quality in VoIP • Low latency, high quality (e.g. NetEQ) jitter buffer necessary • Choose the right codec for the usage scenario • Or a codec that can adapt like iSAC • Transcoding should be avoided, if possible • Significantly better quality than PSTN possible • Wideband coding • No good objective measure for speech quality exists • Always combine with subjective evaluation