<div dir="ltr">Hi everyone<div><br></div><div>I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS with the latest version of Openssl (<font face="arial, sans-serif">'OpenSSL 1.0.1e 11 Feb 2013')</font><br>
<font face="arial, sans-serif">I'm using the default configuration and just uncommentated the '</font> <param name="ws-binding" value=":5066"/> ' in internal.xml in order to have the support for webrtc. </div>
<div>As the client I'm having JSSIP, the latest version with the adjustment to have (<font color="#000000" face="Lucida Grande, sans-serif">DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound:</font></div>
<div><span style="color:rgb(0,0,0);font-family:'Lucida Grande',sans-serif;font-size:13px"><div><br></div><div>[ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)</div><div><br></div></span></div>
<div><font color="#000000" face="Lucida Grande, sans-serif">its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) </font></div><div><font color="#000000" face="Lucida Grande, sans-serif">If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client:</font></div>
<div><font color="#000000" face="Lucida Grande, sans-serif"><div><br></div><div>2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/<a href="http://1000@10.0.14.16:5060">1000@10.0.14.16:5060</a> timer while HOT</div>
<div>2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/<a href="http://1000@10.0.14.16:5060">1000@10.0.14.16:5060</a> Hot Hit 1</div><div><br></div><div>And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts:</div>
<div><div><br></div><div>2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/<a href="http://sip:1000@10.0.14.182:5065">sip:1000@10.0.14.182:5065</a> Hot Hit 4</div><div>2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/<a href="http://sip:1000@10.0.14.182:5065">sip:1000@10.0.14.182:5065</a> timer while HOT</div>
<div>2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)</div></div><div><br></div><div>If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000!</div>
<div><br></div><div>Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? </div><div><br></div><div>Thanks in advanced!</div><div><br></div><div>Sherry</div><div><br></div></font></div>
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