you can look at diffs in the js, but maybe its setting the dtls mandatory?

On Jul 5, 2013, at 1:58 PM, Henry Huang <red.rain.seven@gmail.com> wrote:

I think I have found the issue. If I use the JsSIP website demo to register to webrtc.freeswitch.org then I will be able to replicate the no audio issue on webrtc.freeswitch.org

After testing out different scenarios, it appears to be only when the destination client is the webrtc.freeswitch.org version of JsSIP there will be audio. Neither sipml5 demo or JsSIP demo website registered with webrtc.freeswitch.org can generate audio. Something in the SDP maybe?

Thanks,

Henry


On Wed, Jul 3, 2013 at 9:48 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

The liberal dtmf option will allow all types of dtmf.

On Jul 3, 2013 10:03 PM, "Henry Huang" <red.rain.seven@gmail.com> wrote:
I don't know what it is, but I am getting caught in more issues. I really want to say thank you and it seems like issue on my end since your demo site is working fine. 

I don't think the vanilla config is using dtmf-type "info" by default. I noticed that because my dtmf info start to break. I have from the beginning used dtmf-type info in my internal sip profile and it has been working since April when I dial into 5000 IVR with JsSIP. But since I upgraded to the latest Git last week, I am getting log message saying my channel doesn't support dtmf info, therefore I can't try anything after dialing into the 5000 IVR. 

I am on the verge to reinstall everything and start over. But before I do that, is there anything I am not doing correctly here? Is it Ubuntu? and why would the dtmf info stops working?

Thanks,

Henry


On Wed, Jul 3, 2013 at 7:04 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Its the defaults from the vanilla config with the webrtc opened up.



On Wed, Jul 3, 2013 at 6:09 PM, Henry Huang <red.rain.seven@gmail.com> wrote:
Thanks, the demo site does work. Is it possible that you can share the sip_profile configuration for the demo site here?


On Wed, Jul 3, 2013 at 2:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you should use ext up in 9xxx for chrome too, those lower ext spaces are reserved for other stuff.



On Wed, Jul 3, 2013 at 3:45 PM, Henry Huang <red.rain.seven@gmail.com> wrote:
Anthony,

Thanks, I am able to register a regular SIP to extension 9001 on webrtc.freeswitch.org and I have a user: 4000 on Chrome. When I dial from 9001 to 4000, I get a voice prompt asking for "please enter your ID followed by #" and I don't know how to get pass that.

Thanks,

Henry


On Wed, Jul 3, 2013 at 10:02 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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 range 



On 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? 


Henry


On 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. 
./configure
make
make install
restart freeswitch

This is what I am still seeing. 

2013-07-02 16:40:51.135302 [ALERT] switch_core_media.c:1310 Dispatched RTCP event
2013-07-02 16:40:51.175310 [ALERT] switch_rtp.c:3883 Drop audio packet 70 bytes (dtls not ready!) b=4294967168
2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1510 Setting RTCP src-1 to 10.128.7.103
2013-07-02 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 = -601115572
2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1517 now = 0, now lo = 0, now hi = 0

Henry


On Tue, Jul 2, 2013 at 3:13 PM, Ken Rice <krice@freeswitch.org> wrote:
Did you reconfigure?


On 7/2/13 4:29 PM, "Henry Huang" <red.rain.seven@gmail.com> wrote:

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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