Hi,

We are tracking the new 3.0 release of the mobile SDKs and are curious about this comment in the release notes:
'This is our first WebRTC based release using Chromium WebRTC 57.'

We note that the current version is based around pjsip and as far as I can tell through the pjsip logs it appears that during SDP negotation the Twilio media gateways currently require the use of the G711 codec (a=rtpmap:0 PCMU/8000) which is non-optimal for mobile/wifi networks in particular (QOS configuration notwithstanding).

With the migration to WebRTC does this mean that the Opus codec will now be available and be the preferred/default choice? If so, will we have control over the bitrate, and other codec parameters?

Cheers
Kieron