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, "Joćo 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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_________________________________________________________________________
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http://www.freeswitchsolutions.com
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