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@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@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@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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--

Thanks much,

Noel Morgan
VoiceWARE Communications, Inc.
Direct (214) 390-9113
Mobile (214) 789-7189
visit: http://www.vwci.com/
email: noel@vwci.com


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org