Meetecho

The 19 vs 54 kbps is coherent with the rate they're likely using: I guess PhonerLite is configured to send 16kHz samples, while as also discussed on the Asterisk-dev mailing...

Some more info that may help could be enabling SIP debug in Asterisk. that usually allows for some more details about what leads to a codec being chosen rather than...

I've only given it a quick glance, and found nothing strange about it. I don't have much time to do some testing right now, as soon as I can I'll...

Andrew, that's weird, I don't seem to have any issue with PhonerLite with respect to ICE. I'm using the beta version of PhonerLite, though, which may have fixed something in...

Hi varnav, yes I know, I've been in contact with them, and Opus/VP8 passthrough has been already included in Asterisk 12. Investigations on the legal implications regarging Opus are ongoing,...

The VP8 support you see in Asterisk 12 comes from this patch. Opus is supported as well, but passthrough only, as at the time the Asterisk developers were still concerned...

The warnings you see are not actual warnings, they are debug messages I placed to trace a method I added to trigger a RTCP FIR message. FIR is used to...

This patch doesn't change anything in ICE, and in RTP only adds hooks to the Opus and VP8 codecs, so it's unlikely the cause is the patch itself. Anyway, you...

Ah so it's probably not an ICE/RTP issue, but an SDP one. I guess the patched Asterisk simply is not adding the ICE candidate information. Have you verified it works...

I'm unfamiliar with IAX2, so I'm not sure how codecs are negotiated there. Something that needs to be considered is that Opus always negotiates a 48kHz rate, even when it's...