[Freeswitch-users] OPUS codec use

[Pete Kelly]

Looks like my checked out revision is a few months old.

I'll try the latest 1.2 stable first


On 28 January 2014 15:02, Pete Kelly <pkelly at gmail.com> wrote:

> Noel
>
> That would be useful info. I also don't mind writing stuff up if I get to
> grips with it and its useful.
>
> Initially though I need to understand why FreeSWITCH reports in its logs
> that the sdp is fine but it then zeroes out the port for the actual sdp it
> sends in the SIP
> On 28 Jan 2014 14:55, "Noel Morgan" <noel at vwci.com> wrote:
>
>> Pete,
>>
>> Works great although some of the configuration is hard coded. You can get
>> better performance changing things like complexity.
>>
>> Did I just commit to writing a way to configure this?
>>
>>
>> On Tue, Jan 28, 2014 at 8:27 AM, Pete Kelly <pkelly at gmail.com> wrote:
>>
>>> Hi
>>>
>>> I am trying to use the opus codec built into FreeSWITCH to talk to the
>>> pjsip library (running on an Android handset).
>>>
>>> There seems to be very little docs on opus support within FreeSWITCH
>>> outside the original post saying Anthony has got it working in 2011?
>>>
>>> Does anyone have any experience with using this codec with FreeSWITCH?
>>>
>>>
>>>
>>> _________________________________________________________________________
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>>>
>>> 
>>> 
>>>
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>>>
>>
>>
>> --
>>
>> Thanks much,
>>
>> Noel Morgan
>> VoiceWARE Communications, Inc.
>> Direct (214) 390-9113
>> Mobile (214) 789-7189
>> visit: http://www.vwci.com/
>> email: noel at vwci.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
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>>
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