[Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.

Fri Jul 5 22:10:00 MSD 2013

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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at freeswitch.org> wrote:
>> I think you might need to re bootstrap maybe?
>> 
>> On Jul 2, 2013 8:54 PM, "Henry Huang" <red.rain.seven at 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 at freeswitch.org> wrote:
>> Did you reconfigure? 
>> 
>> 
>> On 7/2/13 4:29 PM, "Henry Huang" <red.rain.seven at 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 at 10.128.7.103 Hot Hit 2
>> 2013-07-02 14:26:47.695312 [ALERT] switch_rtp.c:4425 sofia/internal/1004 at 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 at fusionpbx:/usr/src# openssl version
>> OpenSSL 1.0.1e 11 Feb 2013
>> 
>> 
>> 
>> 
>> On Tue, Jul 2, 2013 at 9:35 AM, Michael Collins <msc at 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 at 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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>> 
>> 
>> 
>> -- 
>> Ken
>> 
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> 
> 
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> 
> 
> -- 
> Anthony Minessale II
> 
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> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
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> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
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> 
> FreeSWITCH Developer Conference
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> 
> _________________________________________________________________________
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> 
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> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
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> AIM: anthm
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> 
> FreeSWITCH Developer Conference
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> pstn:+19193869900
> 
> _________________________________________________________________________
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> 
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