[Freeswitch-users] Significant audio clipping when bridging webrtc to sip

[Joegen Baclor]

Thanks Anthony.  I'm testing without actual early media flowing.  I 
guess I'll just mark this as a known issue with DTLS negotiation for now.

On 02/27/2014 01:14 AM, Anthony Minessale wrote:
> It all depends if the endpoint you are using actually tries to 
> establish media on a 183.  The dtls negotiation needs time.  You 
> should try master instead of 1.4 if you are playing with webrtc 
> because there are many changes going on the 1.4 is probably actually 
> less stable in some cases.
>
> If no media exchange is going on the dtls can't setup so all you can 
> really do is explicitly answer the WebRTC call before bridging to SIP 
> or work on getting media to actually flow during early media stage 
> with the sipml5 guys.
>
>
>
>
>
> On Tue, Feb 25, 2014 at 11:59 PM, Joegen Baclor <jbaclor at ezuce.com 
> <mailto:jbaclor at ezuce.com>> wrote:
>
>     Hi,
>
>     I am using FreeSwitch 1.4-beta to bridge WebRTC->SIP.  All is working
>     great except for the fact that audio is almost always clipped at the
>     beginning of the call.  Looking at the logs, it seems to me that
>     this is
>     due to the delay incurred when performing the DTLS handshake on
>     receipt
>     of media.  I am doing a "pre_answer" before bridging.  My question is,
>     is it possible to do the DTLS handshake prior to originate and right
>     after pre_answer?  Log is in pastebin:
>     http://pastebin.freeswitch.org/22037.  The dropped media happens
>     during
>     the gap between lines 310 and 311.  Thanks for any pointers.
>
>     Joegen
>
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