Hi everyone I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013') I'm using the default configuration and just uncommentated the ' <param name="ws-binding" value=":5066"/> ' in internal.xml in order to have the support for webrtc. As the client I'm having JSSIP, the latest version with the adjustment to have (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: [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) 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: 2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/ 1000@10.0.14.16:5060 timer while HOT 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/ 1000@10.0.14.16:5060 Hot Hit 1 And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts: 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/ sip:1000@10.0.14.182:5065 Hot Hit 4 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/ sip:1000@10.0.14.182:5065 timer while HOT 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) 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! Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? Thanks in advanced! Sherry