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@ezuce.com <mailto:jbaclor@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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