<div dir="ltr"><span style="font-family:arial,sans-serif;font-size:14px">originate user/1001 &echo() works fine with chrome client and the difference is boght/imdroid has no DTLS HANDSHAKE after call answered:</span><br>
<div><span style="font-family:arial,sans-serif;font-size:14px"><br></span></div><div><span style="font-family:arial,sans-serif;font-size:14px"><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><div><br></div></span></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/12/19 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">media_webrtc does resolve this issue, but now there's no audio at both sides, i also tried originate user/1001 &echo() at fs console but no luck.</div>
<div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">
2013/12/19 Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word"><a href="https://wiki.freeswitch.org/wiki/Variable_media_webrtc" target="_blank">https://wiki.freeswitch.org/wiki/Variable_media_webrtc</a><div><br><div><div><div><div>On Dec 18, 2013, at 2:56 AM, Vincent Xia <<a href="mailto:gmangudai@gmail.com" target="_blank">gmangudai@gmail.com</a>> wrote:</div>
<br></div></div><blockquote type="cite"><div><div><div dir="ltr"><div><br></div><div>some updates:</div><div><br></div><div>i found the problem is that boghe/imsdroid sends sip (including REGISTER) over udp, thus taken by fs as a "legacy" endpoint, so the INVITE sent to them contains a "legacy" style sdp whereas they require a "webrtc" style one. i think that's why 488 is generated.</div>
<div><br></div><div>normally the webrtc application takes place between web browsers using sip over websocket, the "legacy" endpoints using sip over udp, but boghe/imsdroid are special, they use sip over udp and supports webrtc, so my question is:</div>
<div><br></div><div>is it possible to have fs send "webrtc" style sdp to sip-over-udp-endpoints? any clue?</div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/12/18 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>codec mismatch is not the problem since i had tried PCMA or OPUS at both ends and fs, now i found this in the webrtc2sip technical guide:</div>
<div><br></div><div>"For example, FreeSWITCH do not support ICE which means it requires the RTCWeb Breaker inorder to be able to connect the browser to a SIP-legacy endpoint."</div>
<div>(in the webrtc2sip webpage it's "For example, if your server doesn't support ICE...", i remember Anthony reminded Mamadou early this year that FreeSWITCH supports ICE so the webpage is updated)</div>
<div><br></div><div>so i guess the webrtc breaker is not architecturally nessesary and the issus may be, say, an incompatibility between fs and doubango clients?</div><div><br></div></div><div><div>
<div class="gmail_extra"><br><br>
<div class="gmail_quote">2013/12/18 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>im testing the interoperability of freeswitch and doubango sip clients including boghe and imsdroid, that both have the setting of media profile: default or webrtc, setting to default the call is fine as from or to the boghe/imdroid client, but setting to webrtc, these clients could only make outgoing calls, when receiving calls they respond the fs invite message with 488 "bad content", the fs console says the call failed due to incompitalbe destination:</div>
<div><br></div><div>2013-12-17 18:03:50.158267 [NOTICE] switch_ivr_originate.c:2699 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]</div><div>2013-12-17 18:03:50.158267 [INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION</div>
<div><br></div><div>the instruction document from doubango says for the webrtc clients to work with the "legacy" sip network a webrtc2sip module is required to as a sip proxy and webrtc breaker, but now freeswitch supports webrtc and sip over websocket, is a webrtc breaker still mandatory?</div>
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