--- title: What Is the Opus Codec? Low-Latency Audio Explained description: Learn what the Opus codec is, how Opus audio delivers low-latency sound, and how it compares with AAC for streaming and calls. image: https://blog.blazingcdn.com/hubfs/Gemini-Blog/image-Oct-24-2026-07-30-20-0452-AM.jpeg --- [![BlazingCDN](https://blog.blazingcdn.com/hubfs/Logo/blog-logo-w.png)](https://blog.blazingcdn.com?hsLang=en-us) [๐Ÿ“˜Learn โ–พ](https://blog.blazingcdn.com/cdn-learn?hsLang=en-us) [CDN Fundamentals](https://blog.blazingcdn.com/cdn-fundamentals?hsLang=en-us) [By Content Type](https://blog.blazingcdn.com/by-content-type?hsLang=en-us) [Advanced Concepts](https://blog.blazingcdn.com/advanced-concepts?hsLang=en-us) [Glossary](https://blog.blazingcdn.com/glossary?hsLang=en-us) [โšก Web Performance](https://blog.blazingcdn.com/web-performance?hsLang=en-us) [๐ŸŽฌVideo & Streaming โ–พ](https://blog.blazingcdn.com/video-streaming-cdn?hsLang=en-us) [๐Ÿ”ด Live Streaming](https://blog.blazingcdn.com/live-streaming?hsLang=en-us) [๐Ÿ“บ VOD & OTT](https://blog.blazingcdn.com/vod-ott?hsLang=en-us) [๐Ÿ’ฐ Bandwidth & Costs](https://blog.blazingcdn.com/bandwidth-costs?hsLang=en-us) [๐Ÿญ Other Industries โ–พ](https://blog.blazingcdn.com/cdn-industry-insights?hsLang=en-us) [๐Ÿ“บ Media & Broadcasting](https://blog.blazingcdn.com/media-broadcasting?hsLang=en-us) [๐Ÿ’พ Software & SaaS](https://blog.blazingcdn.com/software-saas?hsLang=en-us) [๐Ÿ—๏ธ DevOps & Cloud Infra](https://blog.blazingcdn.com/devops-cloud-infra?hsLang=en-us) [๐Ÿ“š AdTech & Advertising](https://blog.blazingcdn.com/adtech-advertising?hsLang=en-us) [๐ŸŽฎ Gaming & Esports](https://blog.blazingcdn.com/gaming-esports?hsLang=en-us) [๐Ÿ“ฑ Mobile Apps & Developers](https://blog.blazingcdn.com/mobile-apps-developers?hsLang=en-us) [๐Ÿค– AI & Machine Learning](https://blog.blazingcdn.com/ai-machine-learning?hsLang=en-us) [๐Ÿ›๏ธ Enterprise & Corporate](https://blog.blazingcdn.com/enterprise-corporate?hsLang=en-us) [๐Ÿ“š E-Learning & EdTech](https://blog.blazingcdn.com/e-learning-edtech?hsLang=en-us) [๐ŸŸ๏ธ Sports & Live Events](https://blog.blazingcdn.com/sports-live-events?hsLang=en-us) [๐Ÿ’ฐ Pricing & Costs โ–พ](https://blog.blazingcdn.com/cdn-pricing-and-cdn-costs?hsLang=en-us) [Provider Pricing](https://blog.blazingcdn.com/provider-pricing?hsLang=en-us) [Cost Optimization](https://blog.blazingcdn.com/cost-optimization?hsLang=en-us) [Decision Support](https://blog.blazingcdn.com/decision-support?hsLang=en-us) [โšกCompare โ–พ](https://blog.blazingcdn.com/cdn-comparison?hsLang=en-us) [Provider Comparisons](https://blog.blazingcdn.com/provider-comparisons?hsLang=en-us) [Strategy Comparisons](https://blog.blazingcdn.com/strategy-comparisons?hsLang=en-us) [Ratings & Benchmarks](https://blog.blazingcdn.com/cdn-ratings-and-benchmarks?hsLang=en-us) [๐Ÿ“Š Benchmarks](https://blog.blazingcdn.com/cdn-ratings-and-benchmarks?hsLang=en-us) [๐Ÿ”’ Security โ–พ](https://blog.blazingcdn.com/cdn-security?hsLang=en-us) [Attack Protection](https://blog.blazingcdn.com/attack-protection?hsLang=en-us) [Encryption & Access](https://blog.blazingcdn.com/encryption-access?hsLang=en-us) [Video & DRM Security](https://blog.blazingcdn.com/video-drm-security?hsLang=en-us) [๐Ÿ“ Case Studies](https://blog.blazingcdn.com/case-studies?hsLang=en-us) [๐Ÿ”Œ Integrations](https://blog.blazingcdn.com/integrations?hsLang=en-us) [๐Ÿ› ๏ธ Tools](https://blog.blazingcdn.com/cdn-tools?hsLang=en-us) [Get Started](https://blazingcdn.com/sign-up-contact-form/) [Learn](https://blog.blazingcdn.com/en-us/tag/learn) [Video - Video & Streaming](https://blog.blazingcdn.com/en-us/tag/video-video-streaming) [Learn - CDN Fundamentals](https://blog.blazingcdn.com/en-us/tag/learn-cdn-fundamentals) # What Is the Opus Codec? Low-Latency Audio Explained Vadym Borovskyi Oct 24, 2026, 9:31:57โ€ฏAM ![](https://blog.blazingcdn.com/hubfs/Gemini-Blog/image-Oct-24-2026-07-30-20-0452-AM.jpeg) The Opus codec is a royalty-free audio codec, standardized by the IETF as RFC 6716 in 2012, that encodes speech and music at 6 to 510 kb/s. Its default algorithmic delay is 26.5 ms, and it can go as low as 5 ms. As of 2026, Opus and G.711 are the two mandatory-to-implement audio codecs in WebRTC, which makes Opus the default audio in browser calls. Opus achieves this range by switching internally between a speech coder and a music coder. A single bitstream can therefore carry a 12 kb/s voice call or 128 kb/s stereo music without renegotiation. ## **How the Opus codec works: SILK, CELT and hybrid mode** Opus combines two older designs. SILK, a linear-prediction coder that came from Skype, handles speech up to wideband. CELT, an MDCT transform coder that came from Xiph.Org, handles fullband audio and music. Hybrid mode runs SILK below 8 kHz and CELT above it, which gives fullband speech at around 24 to 32 kb/s. The encoder can change mode, bandwidth and bitrate on any frame, and every packet's first byte (the TOC byte) tells the decoder how to read it. Timing is set by the frame size. Opus supports 2.5, 5, 10, 20, 40 and 60 ms frames, and a single packet can carry up to 120 ms. Total algorithmic delay is the frame duration plus a fixed lookahead of about 6.5 ms. That gives 26.5 ms at the common 20 ms frame. The restricted low-delay CELT mode removes SILK's extra lookahead and reaches 5 ms at 2.5 ms frames. Three decoder-side features explain why Opus holds up on lossy networks: - **In-band FEC** (low bitrate redundancy, or LBRR) embeds a coarse copy of the previous frame in SILK and hybrid packets, so the decoder can rebuild a single lost packet. - **Packet loss concealment (PLC)** extrapolates audio when no redundancy is available. - **DTX** (discontinuous transmission) sends almost nothing during silence. On a typical call this cuts upstream packets by roughly half. ## **Where Opus audio sits in the stack** Opus produces raw packets, and something else has to carry them. In WebRTC, packets travel over SRTP with the RTP payload format defined in RFC 7587, using a fixed 48 kHz RTP clock whatever the actual coded bandwidth. For files and streaming, Opus is stored in Ogg (RFC 7845), WebM/Matroska, or ISO BMFF (fragmented MP4), which makes it usable in DASH and, with uneven player support, in HLS. Bitrate targets that track the libopus guidance: - Narrowband speech: 8 to 12 kb/s - Wideband speech: 16 to 20 kb/s - Fullband speech: 28 to 40 kb/s - Fullband mono music: 48 to 64 kb/s - Stereo music: 64 to 128 kb/s, which is generally transparent for most listeners at the top of the range ## **The hidden cost of low latency: a worked packet-overhead calculation** Shorter frames cut delay but raise header overhead, and at voice bitrates the overhead dominates. Each SRTP packet over IPv4 carries roughly 50 bytes of headers: 20 bytes IPv4, 8 bytes UDP, 12 bytes RTP and a 10-byte authentication tag. | Frame size | Packets per second | Header overhead | Wire rate at 32 kb/s payload | Algorithmic delay | | --- | --- | --- | --- | --- | | 10 ms | 100 | 40 kb/s | 72 kb/s | 16.5 ms | | 20 ms | 50 | 20 kb/s | 52 kb/s | 26.5 ms | | 40 ms | 25 | 10 kb/s | 42 kb/s | 46.5 ms | Dropping from 20 ms to 10 ms frames buys 10 ms of delay for about 38% more bandwidth, which is why 20 ms remains the WebRTC default. Against the ITU-T G.114 mouth-to-ear target of 150 ms, the 10 ms saving is real but small. A jitter buffer of 20 to 60 ms and a one-way network path of 30 to 100 ms usually matter more. Tune those first. ## **Opus vs AAC and other commonly confused neighbors** **Opus vs. AAC:** AAC-LC is the compatibility default for HLS and DASH and is decoded in hardware on effectively every phone and TV. Its algorithmic delay runs to several tens of milliseconds, and HE-AAC exceeds 100 ms. Public listening tests from the early 2010s put Opus at 64 kb/s on par with or ahead of HE-AAC at the same rate. Opus is also royalty-free, while AAC is licensed through a patent pool. AAC-LD and AAC-ELD reach roughly 15 to 30 ms, but software support for them is thinner. **Opus vs. G.711:** G.711 is uncompressed 64 kb/s narrowband PCM with near-zero codec delay. It remains mandatory in WebRTC for interop with telephony gateways. Opus at 16 to 20 kb/s delivers wideband speech that sounds clearly better, at less than a third of the bitrate. **Opus vs. Ogg:** Ogg is a container, not a codec. A ".opus" file is Opus packets inside Ogg. The same Opus packets can live in WebM, fragmented MP4 or RTP without re-encoding. ## **Configuring Opus for voice: the parameters that matter** In WebRTC, Opus behavior is negotiated in the SDP fmtp line for the Opus payload type. The relevant parameters are: - *useinbandfec=1*: tells the sender that the receiver can use LBRR. Pair it with an encoder packet-loss expectation of 5 to 15% so the encoder actually spends bits on redundancy. - *usedtx=1*: enables discontinuous transmission. Leave it off for music. - *minptime=10* and *ptime*: bound the packetization interval. 20 ms is the safe default. - *maxaveragebitrate*: caps the encoder in bits per second, for example 32000 for fullband speech or 128000 for stereo music. - *stereo=1* and *sprop-stereo=1*: signal that the receiver wants stereo and that the sender will produce it. Browsers default to mono. For file or streaming encodes with libopus, the equivalents are: - Application mode: "voip" for speech, "audio" for music, "lowdelay" for the restricted CELT mode. - Frame duration in milliseconds. - Target bitrate. - Whether VBR is constrained. Constrained VBR keeps packet sizes predictable for segment-based delivery. ## **Common misconceptions about Opus** **"Opus makes live streams low latency."** This is false for HLS and DASH. Segment durations of 2 to 6 seconds, or LL-HLS parts of a few hundred milliseconds, outweigh a 26.5 ms codec delay by one to two orders of magnitude. For segmented delivery, choose the codec on compatibility and bitrate, not latency. **"Opus is only for voice."** CELT mode handles fullband stereo music, and Opus is widely used for music streaming at 96 to 160 kb/s. **"Opus runs at whatever sample rate the RTP clock says."** The 48 kHz in "opus/48000/2" is a fixed signaling convention. The coded bandwidth is chosen dynamically per frame. **"FEC protects against burst loss."** LBRR recovers one lost packet. Bursts need RED (redundant audio), NACK retransmission, or concealment. ## **FAQ: Opus codec, latency and Opus vs AAC** ### Why does WebRTC use the Opus codec? WebRTC uses Opus because it is royalty-free, covers speech and music in one codec, and adds as little as 26.5 ms of algorithmic delay at 20 ms frames. RFC 7874 makes Opus and G.711 mandatory to implement. Opus also brings in-band FEC, DTX and packet loss concealment, which suit unpredictable consumer networks. ### Is Opus better than AAC for live streaming? For interactive WebRTC streams, Opus is the better choice because of its lower delay and royalty-free licensing. For HLS and DASH, AAC-LC is usually the safer pick: hardware decoding is universal and player support is consistent. Opus delivers similar quality at lower bitrates, but its delay advantage disappears behind multi-second segment durations. ### What bitrate should I use for Opus audio? Use 16 to 20 kb/s for wideband speech, 28 to 40 kb/s for fullband speech, and 64 to 128 kb/s for stereo music. These ranges follow the libopus guidance. Below about 12 kb/s, Opus drops to narrowband, so voice stays intelligible but loses clarity. Above roughly 128 kb/s stereo, most listeners hear no further improvement. ### What is the lowest latency the Opus codec can reach? Opus reaches about 5 ms of algorithmic delay using 2.5 ms frames in the restricted low-delay CELT mode. That mode drops SILK, so speech efficiency at low bitrates suffers. Packet header overhead also climbs steeply at 400 packets per second, which is why most deployments stay at 10 or 20 ms frames. ## **Benchmark your audio path and delivery before choosing a codec** This week, split your audio latency budget into its parts: - Log the encoder frame size. - Read jitter buffer delay from WebRTC statistics (the jitterBufferDelay divided by jitterBufferEmittedCount metric). - Compare both against one-way network delay. For segmented streams, measure glass-to-glass delay and attribute it to segment or part duration, not to the codec. Then pick AAC-LC or Opus based on your device matrix rather than on codec latency. WebRTC media is not something BlazingCDN delivers. If your audio ships as pre-encoded HLS, LL-HLS or DASH segments, the [**BlazingCDN HLS Streaming CDN for packaged audio and video segments**](https://blazingcdn.com/streaming-cdn/) serves those segments with a 14-day testing period on real production traffic. 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All rights reserved. ```json { "@context" : "https://schema.org", "@id" : "https://blazingcdn.com/#organization", "@type" : "Organization", "description" : "High-volume CDN for video, live streaming, OTT/IPTV, software, games, SaaS and large-file delivery.", "logo" : { "@id" : "https://blazingcdn.com/#logo", "@type" : "ImageObject", "height" : 560, "url" : "https://blazingcdn.com/wp-content/uploads/2024/08/logo-560-560.png", "width" : 560 }, "name" : "BlazingCDN", "sameAs" : [ "https://www.linkedin.com/company/blazingcdn", "https://x.com/BlazingCdn", "https://www.reddit.com/r/BlazingCDN/", "https://www.wikidata.org/wiki/Q141711074" ], "url" : "https://blazingcdn.com/" } ``` ```json { "@context" : "https://schema.org", "@id" : "https://blog.blazingcdn.com/#website", "@type" : "WebSite", "inLanguage" : "en-US", "name" : "BlazingCDN Blog", "publisher" : { "@id" : "https://blazingcdn.com/#organization" }, "url" : "https://blog.blazingcdn.com/" } ``` ```json { "@context" : "https://schema.org", "@id" : "https://blog.blazingcdn.com/en-us/what-is-the-opus-codec-low-latency-audio-explained#article", "@type" : "BlogPosting", "author" : { "@id" : "https://blog.blazingcdn.com/en-us/author/blazingcdn#person", "@type" : "Person", "description" : "Vadym Borovskyi is Chief Product and Marketing Officer at BlazingCDN, where he has led product strategy and growth since 2021. He has more than 11 years of experience in IT product management and B2B marketing. On this blog he writes about CDN pricing and egress costs, video and software delivery, cache configuration, and how teams evaluate and switch CDN providers. 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Opus delivers similar quality at lower bitrates, but its delay advantage disappears behind multi-second segment durations." }, "name" : "Is Opus better than AAC for live streaming?" }, { "@type" : "Question", "acceptedAnswer" : { "@type" : "Answer", "text" : "Use 16 to 20 kb/s for wideband speech, 28 to 40 kb/s for fullband speech, and 64 to 128 kb/s for stereo music. These ranges follow the libopus guidance. Below about 12 kb/s, Opus drops to narrowband, so voice stays intelligible but loses clarity. Above roughly 128 kb/s stereo, most listeners hear no further improvement." }, "name" : "What bitrate should I use for Opus audio?" }, { "@type" : "Question", "acceptedAnswer" : { "@type" : "Answer", "text" : "Opus reaches about 5 ms of algorithmic delay using 2.5 ms frames in the restricted low-delay CELT mode. That mode drops SILK, so speech efficiency at low bitrates suffers. Packet header overhead also climbs steeply at 400 packets per second, which is why most deployments stay at 10 or 20 ms frames." }, "name" : "What is the lowest latency the Opus codec can reach?" } ] } ```