You can register normal sip clients to that demo too, it has blind reg enabled.Just pick some unique ext numbers up in the 9xxx rangeOn Wed, Jul 3, 2013 at 11:45 AM, Henry Huang <red.rain.seven@gmail.com> wrote:Michael,I understand that webrtc.freeswitch.org works and so is my test server. The one thing I can't verify from webrtc.freeswitch.org is calling from regular SIP client back to webrtc client, and this happened to be the only thing I am having issue with. Everything you see and do on webrtc.freeswitch.org works fine on my test environment.So again, is there anyone who can call from pstn/sip to webrtc without problems?HenryOn Wed, Jul 3, 2013 at 12:58 AM, Michael Jerris <mike@jerris.com> wrote:
yes. you can see it working on the demo at http://webrtc.freeswitch.org.On Jul 3, 2013, at 12:53 AM, Henry Huang <red.rain.seven@gmail.com> wrote:Anyone getting audio working bridging call to webrtc client without bypassing media?
On Jul 2, 2013 6:45 PM, "João Mesquita" <jmesquita@freeswitch.org> wrote:_________________________________________________________________________I think you might need to re bootstrap maybe?
On Jul 2, 2013 8:54 PM, "Henry Huang" <red.rain.seven@gmail.com> wrote:I did, and just in case I ran it again../configuremakemake installrestart freeswitchThis is what I am still seeing.2013-07-02 16:40:51.135302 [ALERT] switch_core_media.c:1310 Dispatched RTCP event2013-07-02 16:40:51.175310 [ALERT] switch_rtp.c:3883 Drop audio packet 70 bytes (dtls not ready!) b=42949671682013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1510 Setting RTCP src-1 to 10.128.7.1032013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1515 Setting RTCP src-1 LENGTH to 12 (1280, 10.128.7.103)2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1516 Setting msw = -1267375797, lsw = -6011155722013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1517 now = 0, now lo = 0, now hi = 0HenryOn Tue, Jul 2, 2013 at 3:13 PM, Ken Rice <krice@freeswitch.org> wrote:Did you reconfigure?Ok, I have upgraded openssl to 1.0.1e, recompiled FreeSWITCH, but I am still getting the "dtls not ready!" alerts and no audio
2013-07-02 14:26:47.695312 [ALERT] switch_rtp.c:4408 sofia/internal/1004@10.128.7.103 Hot Hit 2
2013-07-02 14:26:47.695312 [ALERT] switch_rtp.c:4425 sofia/internal/1004@10.128.7.103 timer while HOT
2013-07-02 14:26:47.715313 [ALERT] switch_rtp.c:5443 Skip sending audio packet 172 bytes (dtls not ready!)
Here is my openssl version:
root@fusionpbx:/usr/src# openssl version
OpenSSL 1.0.1e 11 Feb 2013
On Tue, Jul 2, 2013 at 9:35 AM, Michael Collins <msc@freeswitch.org> wrote:
You'll need to make sure you have OpenSSL 1.0.1e which was released in Feb of this year. I suspect the version in Ubuntu might not be older.
-MC
On Tue, Jul 2, 2013 at 9:02 AM, Henry Huang <red.rain.seven@gmail.com> wrote:
Michael,
I am just using whatever comes with Ubuntu 12.04 64bit.
Package: openssl
Version: 1.0.1-4ubuntu5.8
Depends: libc6 (>= 2.15), libssl1.0.0 (>= 1.0.1)
Thanks,
Henry
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