Dear developers,
I've been redirected to this mailing list by Joshua Colp during fixing a
one way audio bug[1] to discuss another solution as provided in the fix.
Background:
- A lot of people complain about bad VoIP call quality compared to the
old POTS / ISDN devices. What do they mean from a t
Hey Michael,
First off, thanks for taking the time to express some of your thoughts
and concerns to the asterisk-dev list. I'll keep my reply to your
email inline below.
On Mon, Jan 30, 2017 at 4:13 AM, Michael Maier wrote:
> Dear developers,
>
> I've been redirected to this mailing list by Jos
On Mon, Jan 30, 2017 at 3:22 PM, Matt Fredrickson
wrote:
> Hey Michael,
>
> First off, thanks for taking the time to express some of your thoughts
> and concerns to the asterisk-dev list. I'll keep my reply to your
> email inline below.
>
> On Mon, Jan 30, 2017 at 4:13 AM, Michael Maier
> wrote
On Mon, Jan 30, 2017 at 3:32 PM, Matthew Jordan wrote:
>
>
> On Mon, Jan 30, 2017 at 3:22 PM, Matt Fredrickson
> wrote:
>
>> Hey Michael,
>>
>> First off, thanks for taking the time to express some of your thoughts
>> and concerns to the asterisk-dev list. I'll keep my reply to your
>> email in
On Mon, Jan 30, 2017 at 4:44 PM, George Joseph wrote:
>
>
> On Mon, Jan 30, 2017 at 3:32 PM, Matthew Jordan
> wrote:
>
>>
>>
>> On Mon, Jan 30, 2017 at 3:22 PM, Matt Fredrickson
>> wrote:
>>
>>> Hey Michael,
>>>
>>> First off, thanks for taking the time to express some of your thoughts
>>> and
Hello Matt,
thanks for your (and all the other developers) kind response(s)! I'm
happy that you didn't think "please, not one more of this old and
boring discussion". Thanks to all of you taking part regardless!
On 01/30/2017 at 10:22 PM Matt Fredrickson wrote:
> Hey Michael,
>
> First off, tha
Hello Matthew!
On 01/31/2017 at 12:19 AM Matthew Jordan wrote:
> On Mon, Jan 30, 2017 at 4:44 PM, George Joseph wrote:
>
>>
>>
>> On Mon, Jan 30, 2017 at 3:32 PM, Matthew Jordan
>> wrote:
>>
>>>
>>>
>>> On Mon, Jan 30, 2017 at 3:22 PM, Matt Fredrickson
>>> wrote:
>>>
Hey Michael,
>>
On Tue, Jan 31, 2017, at 04:22 AM, Michael Maier wrote:
>
> My first idea was to prevent transcoding already during call setup. But
> now I understand, that this is not a trivial thing to do.
>
> Would it be easier to do it *after* the call has been completely
> established (peer has answered
On 01/31/2017 at 01:38 PM Joshua Colp wrote:
> On Tue, Jan 31, 2017, at 04:22 AM, Michael Maier wrote:
>
>
>
>>
>> My first idea was to prevent transcoding already during call setup. But
>> now I understand, that this is not a trivial thing to do.
>>
>> Would it be easier to do it *after* the ca
On Tue, Jan 31, 2017, at 12:01 PM, Michael Maier wrote:
> >
> > There's no current generic mechanism in the core by which you can
> > request that a reinvite be sent to do this. PJSIP presents a PJSIP
> > specific dialplan function which can do it, and the interface for doing
> > remote RTP bri
On 01/31/2017 at 05:15 PM Joshua Colp wrote:
> On Tue, Jan 31, 2017, at 12:01 PM, Michael Maier wrote:
[...]
>>> We
>>> don't pass the information to the other side, we just adjust our formats
>>> and transcoding.
>>
>> Yes. That's not necessary. But it is necessary, that asterisk is able to
>> i
On Tue, Jan 31, 2017, at 01:29 PM, Michael Maier wrote:
> On 01/31/2017 at 05:15 PM Joshua Colp wrote:
> > On Tue, Jan 31, 2017, at 12:01 PM, Michael Maier wrote:
>
> [...]
>
> >>> We
> >>> don't pass the information to the other side, we just adjust our formats
> >>> and transcoding.
> >>
> >> Y
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