[Freeswitch-users] freeswitch + webrtc + nat, no audio

[Vincent Xia]

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?
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