[Freeswitch-users] Issue with JSSIP + Freeswitch

[Michael Jerris]

We only use non rfc-1918 ip's by default.  If you want to use 1918 ip's you need to tweak acls

Mike


On Aug 5, 2013, at 8:24 AM, Shahrzad A. <shahrzad.aziminia at gmail.com> wrote:

> 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 at 10.0.14.16:5060 timer while HOT
> 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/1000 at 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 at 10.0.14.182:5065 Hot Hit 4
> 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/sip:1000 at 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

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