Hi, I have tried to make a call from Firefox beta 23 to FreeSWITCH and SIP phone connected to FreeSWITCH. It works fine, thanks for the great work!
When calling firefox from FreeSWITCH or SIP phone connected to FreeSWITCH, it gives me no audio. After some investigation, i found that there is crypto tag in SDP from freeswitch. I guess it might be one of the reasons for no audio issue. Any idea? Another question, i can start chrome from command line to enable webrtc debugging. Can i do the same with firefox? How? v=0 o=FreeSWITCH 1372939317 1372939318 IN IP4 103.11.143.70 s=FreeSWITCH c=IN IP4 103.11.143.70 t=0 0 a=msid-semantic: WMS 2BsFTdOGTL8vmTeIjsOBD3G1hm9po8ag m=audio 18820 RTP/SAVPF 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fingerprint:sha-256 54:B4:FA:C5:31:B2:AE:0D:64:AF:41:E5:D0:AC:1A:E9:70:9C:7C:52:95:0A:3C:7A:4C:FF:8B:B4:5F:0B:EA:4D a=rtcp-mux a=rtcp:18820 IN IP4 103.11.143.70 a=ssrc:1419290113 cname:zsMivrPf2OBfyFOI a=ssrc:1419290113 msid:2BsFTdOGTL8vmTeIjsOBD3G1hm9po8ag a0 a=ssrc:1419290113 mslabel:2BsFTdOGTL8vmTeIjsOBD3G1hm9po8ag a=ssrc:1419290113 label:2BsFTdOGTL8vmTeIjsOBD3G1hm9po8aga0 a=ice-ufrag:XsarlGitThI7fIdF a=ice-pwd:1QOn7us09U5xqJ5e a=candidate:1642238858 1 udp 659136 103.11.143.70 18820 typ host generation 0 a=candidate:1642238858 2 udp 659136 103.11.143.70 18820 typ host generation 0 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9bT3HoWcsokFaI/XVDhSTxLuuGIoWOeyQeDlVLKL a=ptime:20 -- Iwan Budi Kusnanto _______________________________________________ dev-media mailing list [email protected] https://lists.mozilla.org/listinfo/dev-media
