HD voice and wideband audio end to end

See also

Sources

1. 1
   The Cisco Wireless Phone 9821 uses G.722 instead of G.711 only when Unified CM is configured for G.722 and the far endpoint supports G.722, regardless of whether a wideband headset or handset is enabled.

   Set up wideband codec (Cisco Wireless Phone 9821) · Set up wideband codec, introductory paragraph · Checked 2026-10-14
2. 2
   On the Cisco Wireless Phone 9821 the Advertise G.722 and iSAC Codec product-specific parameter can be set to Enabled or Disabled to override the enterprise parameter.

   Set up wideband codec (Cisco Wireless Phone 9821) · Set up wideband codec, product-specific parameter step · Checked 2026-10-14
3. 3
   RFC 7875 notes that many legacy mobile terminals support only narrowband AMR and recommends offering AMR alongside AMR-WB with AMR-WB listed first.

   RFC 7875: Additional WebRTC Audio Codecs for Interoperability · Sections 4.1.3 and 4.2.2 · Checked 2026-10-14
4. 4
   Offering AMR-WB lets a WebRTC endpoint reach 3GPP mobile wideband terminals without transcoding.

   RFC 7875: Additional WebRTC Audio Codecs for Interoperability · Section 4.1.2 · Checked 2026-10-14
5. 5
   AMR-WB is very widely implemented in mobile devices and networks offering HD voice, including VoLTE.

   RFC 7875: Additional WebRTC Audio Codecs for Interoperability · Section 4.1.2 · Checked 2026-10-14
6. 6
   AMR-WB is registered as media type audio/AMR-WB and appears in SDP as AMR-WB/16000.

   RFC 4867: RTP Payload Format and File Storage Format for the AMR and AMR-WB Audio Codecs · Section 8.2 and Section 8.3.3 SDP examples · Checked 2026-10-14
7. 7
   AMR-WB has nine wideband codec modes ranging from 6.6 to 23.85 kbit/s.

   RFC 4867: RTP Payload Format and File Storage Format for the AMR and AMR-WB Audio Codecs · Section 3.2 · Checked 2026-10-14
8. 8
   Unlike G.722, AMR-WB signals its true rate: the SDP rtpmap clock rate must be 16000 for AMR-WB and the RTP timestamp advances 320 per 20 ms frame.

   RFC 4867: RTP Payload Format and File Storage Format for the AMR and AMR-WB Audio Codecs · Section 4.1 and Section 8.2 (media type registration) · Checked 2026-10-14
9. 9
   AMR-WB samples at 16000 Hz, giving 320 samples per 20 ms frame.

   RFC 4867: RTP Payload Format and File Storage Format for the AMR and AMR-WB Audio Codecs · Section 3.2 · Checked 2026-10-14
10. 10
    In a unicast SDP answer, each accepted m= line must contain at least one media format from the offer, and the answerer is recommended to keep the offer's relative order.

    RFC 3264 — An Offer/Answer Model with the Session Description Protocol (SDP) · Section 6.1 · Checked 2026-10-14
11. 11
    The SDP answerer must send using a format from the offer that is also in the answer and should use the offer's most preferred such format.

    RFC 3264 — An Offer/Answer Model with the Session Description Protocol (SDP) · Section 6.1 · Checked 2026-10-14
12. 12
    In Unified CM the Advertise G.722 Codec enterprise parameter defaults to Enabled, so registered Cisco IP phones advertise G.722.

    Set up wideband codec (Cisco Wireless Phone 9821) · Set up wideband codec, enterprise parameter step · Checked 2026-10-14
13. 13
    Under RFC 3264 the codec actually used on a SIP leg follows the offer's preference order among commonly supported formats, so a device that lists G.711 ahead of a wideband codec will usually get narrowband even when both ends support wideband.inferred

    RFC 3264 — An Offer/Answer Model with the Session Description Protocol (SDP) · Sections 5.1, 6.1 and 7 · Checked 2026-10-14
14. 14
    Once any media leg in a call path is negotiated to G.711, the audio delivered end to end is limited to narrowband, because no later hop can restore frequencies the G.711 leg did not carry.inferred

    Plan for media bypass with Direct Routing · Section Use of Media Processors and Transport Relays, transcoding explanation (SILK to G.711 example) · Checked 2026-10-14
15. 15
    Because G.722 is signalled with an 8000 Hz RTP clock while sampling at 16000 Hz, an implementation that derives timestamps from the true sample rate will misstate RTP timing for G.722 streams.inferred

    RTP Profile for Audio and Video Conferences with Minimal Control · Section 4.5.2 G722 · Checked 2026-10-14
16. 16
    G.722 is the wideband codec required for DECT CAT-iq terminals.

    RFC 7875: Additional WebRTC Audio Codecs for Interoperability · Section 4.3.2 · Checked 2026-10-14
17. 17
    RFC 7875 describes G.722 as a royalty-free low-complexity ITU-T wideband codec whose support allows transcoding-free calls with fixed VoIP terminals.

    RFC 7875: Additional WebRTC Audio Codecs for Interoperability · Section 4.3 (4.3.1 and 4.3.2) · Checked 2026-10-14
18. 18
    The RTP clock rate signalled for G.722 is 8000 Hz rather than 16000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility.

    RTP Profile for Audio and Video Conferences with Minimal Control · Section 4.5.2 G722 · Checked 2026-10-14
19. 19
    G.722 audio is actually sampled at 16,000 Hz.

    RTP Profile for Audio and Video Conferences with Minimal Control · Section 4.5.2 G722 · Checked 2026-10-14
20. 20
    G.722 is assigned static RTP payload type 9 with a clock rate of 8,000 and one channel.

    RTP Profile for Audio and Video Conferences with Minimal Control · Section 6 Table 4 (payload type 9 row) · Checked 2026-10-14
21. 21
    ITU-T G.722 is titled 7 kHz audio-coding within 64 kbit/s, and the in-force edition is G.722 (09/12) with Amendment 1 (10/14).

    G.722: 7 kHz audio-coding within 64 kbit/s · Recommendation landing page, in-force list · Checked 2026-10-14
22. 22
    When the offerer sends media, it should use the first media format listed in the answer.

    RFC 3264 — An Offer/Answer Model with the Session Description Protocol (SDP) · Section 7 · Checked 2026-10-14
23. 23
    RFC 7587 defines narrowband as 0-4000 Hz audio at 8000 Hz sampling and wideband as 0-8000 Hz audio at 16000 Hz sampling, alongside mediumband, super-wideband (0-12000 Hz) and fullband (0-20000 Hz).

    RFC 7587: RTP Payload Format for the Opus Speech and Audio Codec · Section 2 Table 1 · Checked 2026-10-14
24. 24
    Opus scales from 6 kbit/s narrowband mono speech to 510 kbit/s fullband stereo music.

    RFC 6716: Definition of the Opus Audio Codec · Section 2 · Checked 2026-10-14
25. 25
    Opus can switch seamlessly between its operating modes and audio bandwidths mid-stream without renegotiating the session, and any decoder can decode any bandwidth.

    RFC 6716: Definition of the Opus Audio Codec · Section 2 and Section 2.1.3 · Checked 2026-10-14
26. 26
    Enabling Twilio's Limited Availability Opus codec does not by itself establish wideband audio to a PSTN callee, because Twilio says LA codecs are not supported end to end.inferred

    Elastic SIP Trunking Codecs · Limited Availability codecs section · Checked 2026-10-14
27. 27
    The Opus maxplaybackrate parameter is a receiver hint of the highest output sampling rate it can play (8000 to 48000, default 48000); encoders should not encode above it but decoders must still decode any bandwidth.

    RFC 7587: RTP Payload Format for the Opus Speech and Audio Codec · Section 6.1 and Section 7.1 · Checked 2026-10-14
28. 28
    The Opus RTP timestamp uses a 48000 Hz clock for all Opus modes and sampling rates.

    RFC 7587: RTP Payload Format for the Opus Speech and Audio Codec · Section 4.1 · Checked 2026-10-14
29. 29
    Opus is always signalled in SDP as opus/48000/2, even for mono sessions; mono or stereo preference is carried by the stereo parameter.

    RFC 7587: RTP Payload Format for the Opus Speech and Audio Codec · Section 7 and Section 7.1 Example 1 · Checked 2026-10-14
30. 30
    Opus uses its SILK layer for narrowband to wideband speech, its CELT layer for up to fullband, and a hybrid mode for super-wideband and fullband in which SILK codes up to 8 kHz and CELT codes above.

    RFC 6716: Definition of the Opus Audio Codec · Section 2 · Checked 2026-10-14
31. 31
    The Opus sprop-maxcapturerate parameter is a sender hint of the highest input sampling rate it is likely to produce, not a guarantee, with range 8000 to 48000 and default 48000.

    RFC 7587: RTP Payload Format for the Opus Speech and Audio Codec · Section 6.1 and Section 7.1 · Checked 2026-10-14
32. 32
    The SDP rtpmap clock rate is not a reliable indicator of audio bandwidth: G.722 shows 8000 although wideband, and Opus shows 48000 whatever bandwidth is actually encoded.inferred

    RFC 7587: RTP Payload Format for the Opus Speech and Audio Codec · Sections 4.1 and 7 (Opus); read with RFC 3551 Section 4.5.2 · Checked 2026-10-14
33. 33
    AMR-WB on the Teams Direct Routing SBC leg is supported for non-media-bypass calls only.

    Plan Direct Routing · Section Supported codecs (AMR-WB list item) · Checked 2026-10-14
34. 34
    With media bypass the SBC must itself negotiate a codec the Teams client supports, since no Microsoft component transcodes on that path; AMR-WB is not available there.inferred

    Plan for media bypass with Direct Routing · Section Use of Media Processors and Transport Relays (table row Can do transcoding) · Checked 2026-10-14
35. 35
    Teams Direct Routing supports SILK, G.711, G.722, G.729 and AMR-WB between Microsoft Teams and the SBC.

    Plan Direct Routing · Section Supported codecs · Checked 2026-10-14
36. 36
    The Plan Direct Routing supported-codecs list does not include Opus.

    Plan Direct Routing · Section Supported codecs · Checked 2026-10-14
37. 37
    On a media-bypass trunk Teams still inserts Media Processors when a call escalates to a group call, goes to a federated Teams user, or is forwarded or transferred to a Skype for Business user.

    Plan for media bypass with Direct Routing · Section Use of Teams Media Processors if trunk is configured for media bypass · Checked 2026-10-14
38. 38
    The Teams Media Processor is a B2BUA that can change codecs, for example SILK between Teams client and Media Processor and G.711 between Media Processor and SBC.

    Plan for media bypass with Direct Routing · Section Use of Media Processors and Transport Relays, transcoding explanation footnote · Checked 2026-10-14
39. 39
    Teams Media Processors are always in the media path for voice applications such as Call Park, Auto Attendant and Call Queues, and for Teams web clients, even on media-bypass trunks.

    Plan for media bypass with Direct Routing · Sections Use of Media Processors and Transport Relays; Requirements for using media processors · Checked 2026-10-14
40. 40
    A non-bypassed Teams Direct Routing call whose SBC leg negotiates G.711 is narrowband end to end even though the Teams client leg uses SILK.inferred

    Plan for media bypass with Direct Routing · Section Use of Media Processors and Transport Relays, transcoding explanation · Checked 2026-10-14
41. 41
    Teams Transport Relays are not B2BUAs, so the codec between Teams client and SBC is never changed when a bypassed call flows through a relay.

    Plan for media bypass with Direct Routing · Section Use of Media Processors and Transport Relays, comparison table and transcoding explanation · Checked 2026-10-14
42. 42
    The Twilio Elastic SIP Trunking codecs page lists no G.722 support, in either the GA or the Limited Availability set.

    Elastic SIP Trunking Codecs · Whole page (GA and Limited Availability lists) · Checked 2026-10-14
43. 43
    Twilio Elastic SIP Trunking supports only PCMU and PCMA by default (generally available).

    Elastic SIP Trunking Codecs · Opening section of Elastic SIP Trunking Codecs · Checked 2026-10-14
44. 44
    A call carried over a Twilio Elastic SIP Trunk with only the default GA codecs is narrowband on the trunk leg, since only PCMU and PCMA are available.inferred

    Elastic SIP Trunking Codecs · Opening section (GA codec list) · Checked 2026-10-14
45. 45
    Twilio states that Limited Availability codecs are not supported end to end on Elastic SIP Trunking.

    Elastic SIP Trunking Codecs · Limited Availability codecs section · Checked 2026-10-14
46. 46
    Twilio offers G.729, Opus and AMR-NB on Elastic SIP Trunking only as Limited Availability codecs requested through sales or the account team.

    Elastic SIP Trunking Codecs · Limited Availability codecs section · Checked 2026-10-14
47. 47
    With Limited Availability codecs enabled, Twilio origination SDP lists codecs in the order Opus, PCMU, PCMA, G.729, AMR-NB; without them it sends PCMU then PCMA.

    Elastic SIP Trunking Codecs · Origination section · Checked 2026-10-14
48. 48
    For termination, Twilio Elastic SIP Trunking matches the first supported codec in the customer's SDP offer.

    Elastic SIP Trunking Codecs · Termination section · Checked 2026-10-14
49. 49
    WebRTC endpoints are required to implement Opus and G.711 (PCMA and PCMU); no wideband codec other than Opus is mandatory.

    RFC 7874: WebRTC Audio Codec and Processing Requirements · Section 3 · Checked 2026-10-14
50. 50
    RFC 7874 recommends that an endpoint able to process audio above 8 kHz offer Opus before PCMA and PCMU.

    RFC 7874: WebRTC Audio Codec and Processing Requirements · Section 3 · Checked 2026-10-14
51. 51
    Webex Calling states each audio stream consumes 70-80 kbps and recommends planning 100 kbps per stream, about 200 kbps for a two-stream internal call.

    Audio specifications for Webex Calling · Bandwidth section · Checked 2026-10-14
52. 52
    Analog telephone adapters and DECT phones do not support Opus on Webex Calling.

    Audio specifications for Webex Calling · Codec section, device support note · Checked 2026-10-14
53. 53
    Webex Calling does not support calls offering only G.729, and G.711 must always be included in invites.

    Audio specifications for Webex Calling · Codec section, G.729 note · Checked 2026-10-14
54. 54
    Webex Calling uses Opus for audio where supported, including calls between Webex App and Multiplatform Phones and calls to Auto Attendant and Voicemail.

    Audio specifications for Webex Calling · Codec section of Audio specifications for Webex Calling · Checked 2026-10-14
55. 55
    Because most PSTN providers do not support Opus, Webex Calling calls over the PSTN use G.711.

    Audio specifications for Webex Calling · Codec section, PSTN note · Checked 2026-10-14

Documents

Cite this page

WarmTransfer. (2026, October 14). HD voice and wideband audio end to end. WarmTransfer. https://warmtransfer.net/knowledge/hd-voice-wideband

@misc{warmtransfer-hd-voice-wideband,
  title  = {HD voice and wideband audio end to end},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/knowledge/hd-voice-wideband},
  note   = {Verified 2026-10-14}
}