<div dir="ltr">issue resolve by upgrading chrome at nat side from 28 to 31.</div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/12/5 Vincent Xia <span dir="ltr"><<a href="mailto:gmangudai@gmail.com" target="_blank">gmangudai@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><br></div><div>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:</div>
<div><br></div><div>[ERR] switch_rtp.c:2506 audio DTLS packet not written</div><div>[ERR] switch_rtp.c:2506 audio DTLS packet not written</div><div>[ERR] switch_rtp.c:2506 audio DTLS packet not written</div><div>[ERR] switch_rtp.c:2506 audio DTLS packet not written</div>
<div>[ERR] switch_rtp.c:2506 audio DTLS packet not written</div><div>......</div><div><br></div><div>when making call between 2 parties(A-LAN-FS-NAT-B), there's no audio between the 2 parties,the fs log looks fine:</div>
<div>[NOTICE] switch_rtp.c:1074 Auto Changing stun/rtp/dtls port from ex_ip:57777 to in_ip:57777</div><div>[INFO] switch_rtp.c:2449 Changing audio DTLS state from HANDSHAKE to SETUP</div><div>[INFO] switch_rtp.c:2368 audio Fingerprint Verified.</div>
<div>[INFO] switch_rtp.c:2868 Activating Audio Secure RTP SEND</div><div>[INFO] switch_rtp.c:2846 Activating Audio Secure RTP RECV</div><div>[DEBUG] switch_core_sqldb.c:2357 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80</div>
<div>[INFO] switch_rtp.c:2408 Changing audio DTLS state from SETUP to READY</div><div>[DEBUG] switch_core_sqldb.c:2357 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80</div><div>[DEBUG] switch_rtp.c:1719 RTCP packet not written</div>
<div><br></div><div>but the chrome console of party B says:</div><div>JsSIP | MEDIA SESSION | Peerconnection status: undefined</div><div>JsSIP | MEDIA SESSION | Ice Status: undefined</div><div>JsSIP | MEDIA SESSION | No more ICE candidate</div>
<div>JsSIP | MEDIA SESSION | Peerconnection status: undefined</div><div>JsSIP | MEDIA SESSION | Ice Status: undefined</div><div>JsSIP | MEDIA SESSION | No more ICE candidate</div><div>...</div><div><br></div><div>could anybody give some help on this?</div>
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