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

Anthony Minessale anthony.minessale at gmail.com
Wed Feb 26 20:14:40 MSK 2014

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

On Tue, Feb 25, 2014 at 11:59 PM, Joegen Baclor <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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