WebRTC call audio delay

Hi all I testing latest asterisk 13.15.0 and Chrome 57.
Еverything looks ok. But When looking at the CLI traces I see immediately upon answering and echo.
Apr 12 16:11:48] DEBUG[20244][C-0000001f]: chan_sip.c:28743 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[Apr 12 16:11:48] DEBUG[20440][C-0000001f]: channel.c:2845 __ast_answer: Didn’t receive a media frame from SIP/8900-0000000f within 500 ms of answering. Continuing anyway
[Apr 12 16:11:48] DEBUG[20440][C-0000001f]: pbx.c:2875 pbx_extension_helper: Launching ‘Echo’
– Executing [9888@outgoing:3] Echo(“SIP/8900-0000000f”, “”) in new stack
[Apr 12 16:11:56] DEBUG[20440][C-0000001f]: res_srtp.c:516 ast_srtp_add_stream: Adding new policy for SSRC 442472126
[Apr 12 16:11:56] DEBUG[20440][C-0000001f]: res_rtp_asterisk.c:4659 ast_rtp_read: 0x7fa2f803fdd0 – Probation learning mode pass with source address 194.186.195.46:52697
> 0x7fa2f803fdd0 – Probation passed - setting RTP source address to 194.186.XX.YY:52697
So i have 8 sec delay after Echo and media probation. How i can fix it?