On Mar 15, 2011 11:32 AM, "Alastair Johnson" <
address@hidden> wrote:
> On Tuesday 15 March 2011, Dragos D wrote:
>> > Another approach would be a general instruction to start at the minimum,
>> > and
>> > increase until quality is good enough. If you step up and the picture
>> > breaks
>> > up then step down again , as that's as quick as your connection can
>> > handle.
>> >
>> > For the future we could look at giving some diagnostic feedback and
>> > advice if
>> > the datastream is lossy or can't keep up. Perhaps an indicator that you
>> > can click to reduce bitrate and reconnect the call? I guess if it was
>> > reasonable
>> > to approximate adaptive bitrate through reinvites everyone would be doing
>> > it.
>>
>> These are some of the reasons for which, in general, Skype is a better
>> solution. It can change on the fly the bitrate and adapt the image
>> parameters accordingly. I say "in general" because sometimes I was not
>> satisfied with their choices. They can adapt the resolution, framerate...
>> to a chosen bitrate ( determined upon analyzing the network conditions).
>> It can actually do 640x480 at 40 kbits/sec (if I remember correctly) !!!
>> This I something I could not do with h.264 or any other available codec.
>
> I gave up on Skype when it couldn't reliably do 2-way audio through my
> firewall, and there was no information on how to fix it because it's supposed
> to "Just Work (tm)". Getting SIP working took some doing, but it's
> sufficiently open that it was possible if I put the effort in. The target
> should be to get the best of both; something that usually Just Works, but that
> is fixable on the rare occasions that it doesn't, and can be tweaked to better
> suit specific situations by anyone sufficiently clued up.
>
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