<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">We only use non rfc-1918 ip's by default. If you want to use 1918 ip's you need to tweak acls<div><br></div><div>Mike</div><div><br></div><div><br><div><div>On Aug 5, 2013, at 8:24 AM, Shahrzad A. <<a href="mailto:shahrzad.aziminia@gmail.com">shahrzad.aziminia@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><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 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="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 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 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 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></font></div></div></blockquote></div><br></div></body></html>