to test freeswitch webrtc with chrom + jssip, using the latest git version 1.5.8b to build and run under windows, when calling an example ivr(e.g. 5000 or 9386) from inside the same LAN(A-LAN-FS), everything is fine, but calling from behind nat(FS-NAT-B), i got abundant errors on the fs console after the call get answered:

[ERR] switch_rtp.c:2506 audio DTLS packet not written

[ERR] switch_rtp.c:2506 audio DTLS packet not written

[ERR] switch_rtp.c:2506 audio DTLS packet not written

[ERR] switch_rtp.c:2506 audio DTLS packet not written

[ERR] switch_rtp.c:2506 audio DTLS packet not written

......

when making call between 2 parties(A-LAN-FS-NAT-B), there's no audio between the 2 parties,the fs log looks fine:

[NOTICE] switch_rtp.c:1074 Auto Changing stun/rtp/dtls port from ex_ip:57777 to in_ip:57777

[INFO] switch_rtp.c:2449 Changing audio DTLS state from HANDSHAKE to SETUP

[INFO] switch_rtp.c:2368 audio Fingerprint Verified.

[INFO] switch_rtp.c:2868 Activating Audio Secure RTP SEND

[INFO] switch_rtp.c:2846 Activating Audio Secure RTP RECV

[DEBUG] switch_core_sqldb.c:2357 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80

[INFO] switch_rtp.c:2408 Changing audio DTLS state from SETUP to READY

[DEBUG] switch_core_sqldb.c:2357 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80

[DEBUG] switch_rtp.c:1719 RTCP packet not written

but the chrome console of party B says:

JsSIP | MEDIA SESSION | Peerconnection status: undefined

JsSIP | MEDIA SESSION | Ice Status: undefined

JsSIP | MEDIA SESSION | No more ICE candidate

JsSIP | MEDIA SESSION | Peerconnection status: undefined

JsSIP | MEDIA SESSION | Ice Status: undefined

JsSIP | MEDIA SESSION | No more ICE candidate

...

could anybody give some help on this?