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

[Vincent Xia]

issue resolve by upgrading chrome at nat side from 28 to 31.


2013/12/5 Vincent Xia <gmangudai at 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?
>
>
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