issue resolve by upgrading chrome at nat side from 28 to 31. 2013/12/5 Vincent Xia <gmangudai@gmail.com>
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?