Last Call Review of draft-ietf-rtcweb-audio-codecs-for-interop-05
review-ietf-rtcweb-audio-codecs-for-interop-05-opsdir-lc-liu-2016-03-15-00

Hi all,



I have reviewed draft-ietf-rtcweb-audio-codecs-for-interop-05 as part of the
Operational directorate's ongoing effort to review all IETF documents being
processed by the IESG.  These comments were written with the
 intent of improving the operational aspects of the IETF drafts. Comments that
 are not addressed in last call may be included in AD reviews during the IESG
 review.  Document editors and WG chairs should treat these comments just like
 any other last call comments.



“This document provides some guidelines on the suitable codecs to be 
considered for WebRTC endpoints to address the most relevant  interoperability
use cases.”



My overall view of the document is 'Ready with nits' for publication.



**** Technical ****



Question: I am not sure why this I-D is on the Informational track rather than
BCP or Std Track.







**** Editorial ****



* Section 2, page 3:

>

>    o  Legacy networks: In this document, legacy networks encompass the

>       conversational networks that are already deployed like the PSTN,

>       the PLMN, the IP/IMS networks offering VoIP services, including

>       3GPP "4G" Evolved Packet System[TS23.002]



Missing space in "Evolved Packet System[TS23.002]"





* Section 2, page 3:

>  o  PSTN:Public Switched Telephone Network



Missing space.





* Section 3, page 4:

>  Consequently,

>    a significant number of calls are likely to occur between terminals

>    supporting WebRTC endpoints and other terminals like mobile handsets,

>    fixed VoIP terminals, DECT terminals that do not support WebRTC

>    endpoints nor implement OPUS.



Seems should  s/terminals, DECT terminals/terminals, and DECT terminals/





* Section 3: each of the bullets is separated by two blank lines rather than a
single one.





* Section 4.1.1, page 5:

> especially



s/especially/specially/





* Section 4.1.3, page 5:

>    The payload format to be used for AMR-WB is described in [RFC4867]

>    with bandwidth efficient format and one speech frame encapsulated in

>    each RTP packets



s/packets/packet/





* Section 4.2.1, page 6:

>  This include both mobile phone calls using GSM and 3G



s/include/includes/





* Section 4.2.1, page 6:

> such as, GSMA voice IMS profile for VoLTE in [IR.92].



Please remove the comma.





* Section 4.2.1, page 6:

>    degrading the high efficiency over mobile radio access.References

> for



Missing space.





* Section 4.2.3, page 7:

>    The payload format to be used for AMR is described in [RFC4867] with

>    bandwidth efficient format and one speech frame encapsulated in each

>    RTP packets.



s/packets/packet/



Regards,

Will (Shucheng LIU)