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

Tue Jul 2 19:20:51 MSD 2013

what version of openssl are you building against?

On Jul 2, 2013, at 4:37 AM, Ashley B <ashley at midletearth.com> wrote:

> Henry
>  
> Thank you for pointing this out, we are experiencing the same issue (fsctl debug_level 10)
>  
> 2013-07-02 08:13:27.999216 [ALERT] switch_rtp.c:3877 Drop audio packet 182 bytes (dtls not ready!) b=4294967168
> 2013-07-02 08:13:27.999216 [ALERT] switch_rtp.c:5437 Skip sending audio packet 13 bytes (dtls not ready!)
>  
> Rgds
> Ashley.
>  
>  
> From: Henry Huang [mailto:red.rain.seven at gmail.com] 
> Sent: Monday, July 1, 2013 11:08 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
>  
> I clearly don't understand why this is not working. Have Chrome dial 5000 for the demo IVR works. The only time it wouldn't work is when any kind of client is trying to dial a WebRTC client, that's when audio stops working and I get this "dtls not ready" thing in the log. What steps should I take to make dtls ready?
>  
> I have grabbed level 10 debug with SIP trace here if that helps:
>  
> http://pastebin.freeswitch.org/21138
>  
> Thanks,
>  
> Henry
>  
> 
> On Mon, Jul 1, 2013 at 9:09 AM, Henry Huang <red.rain.seven at gmail.com> wrote:
> Hi Anthony,
>  
> Thanks for the input. I suspect it to be some kind of firewall in my network preventing me from getting the Stun request back. Does the ext-rtp-ip use a default stun server? can I change the default stun server that it uses? or should I just try to use the stun server lookup with "stun:stun.server.com"?
>  
> Thanks,
>  
> Henry
>  
> 
> On Fri, Jun 28, 2013 at 4:45 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> You really should have enabled the sip trace too in your pastebin.
>  
> until wireshark catches up with webrtc diagnosing stuff like this is a little more difficult.
>  
> It looks like the address your FS is telling to the browser to send to is unreachable because it has not received any of the stun or dtls packets from the candidates.
>  
> Try answering the call in your dialplan before you bridge and use {ignore_early_media=true} on your dial string.  The ice stuff does not like waiting and I am also not sure if jssip has implemented early media.
>  
> You could compare results with sipml5 and you can also contact the user groups for both projects on google groups for additional insight.
>  
>  
>  
>  
>  
>  
>  
>  
>  
> 
> On Fri, Jun 28, 2013 at 4:50 PM, Henry Huang <red.rain.seven at gmail.com> wrote:
> bypass_media is commented out. Here is the level 10 debug pastebin. I tried it with both end as Chrome client and still no audio. The working model are the following with bypass_media=false:
> Chrome => Softphone     (2 way audio good)
> Softphone => Chrome     (No Audio)
> Chrome => Chrome        (No Audio)
>  
> http://pastebin.freeswitch.org/21129
>  
> Thanks,
>  
> Henry
>  
> 
> On Fri, Jun 28, 2013 at 12:11 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Check if you have bypass_media set, you cannot use bypass media between a soft-phone and webrtc enabled endpoints.
> That is not a webrtc SDP so if that's not your problem you need a more complete log to examine.
>  
>  
> 
> On Fri, Jun 28, 2013 at 1:56 PM, Henry Huang <red.rain.seven at gmail.com> wrote:
> Hi, 
>  
> I am getting no audio when calling from regular sip client to Chrome/JSsip. FreeSWITCH version is the latest (Version 1.5.3b git a52a604 2013-06-28 16:05:27Z)
> And JsSIP directly from their demo site. 
>  
> Audio works if I am calling from browser to softphone. But no audio at all if calling back from softphone to browser. What kind of information would be good to help debug this?
>  
> 2013-06-28 11:52:01.271125 [NOTICE] sofia.c:6517 Channel [sofia/internal/sip:9ku10lcl at 5g4f1tqaruji.invalid] has been answered
> 2013-06-28 11:52:01.271125 [DEBUG] switch_channel.c:3595 (sofia/internal/sip:9ku10lcl at 5g4f1tqaruji.invalid) Callstate Change RINGING -> ACTIVE
> 2013-06-28 11:52:01.291128 [DEBUG] switch_core_codec.c:244 sofia/internal/1003 at 72.1.46.122Restore previous codec PCMU:0.
> 2013-06-28 11:52:01.291128 [DEBUG] mod_sofia.c:822 Local SDP sofia/internal/1003 at 72.1.46.122:
> v=0
> o=FreeSWITCH 1372428618 1372428620 IN IP4 72.1.46.122
> s=FreeSWITCH
> c=IN IP4 72.1.46.122
> t=0 0
> m=audio 16892 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>  
>  
> Thanks,
>  
> Henry
>  
> 
> On Fri, Jun 28, 2013 at 7:59 AM, Gerald Weber <gerald.weber at besharp.at> wrote:
> Hi,
>  
> try the following in your pastebin http://pastebin.freeswitch.org/21123
>  
> Remove Line 11 and put this after line 18 (var rtcSession = e.sender;)
>  
> var remoteView = $('#remoteView‘);
>  
>  
> does it work then ?
>  
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von João Mesquita
> 
> Gesendet: Freitag, 28. Juni 2013 00:57
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
>  
> Tony, here's my log:
> http://pastebin.freeswitch.org/21122
>  
> Here's the html page I am using:
> http://pastebin.freeswitch.org/21123
>  
> I am pretty certain that I must be doing something awfully wrong on the JSSIP part or on the config because it simply works with the JSSIP tryit...
>  
> Let me know how else I can help.
>  
> 
> João Mesquita
> FreeSWITCH™ Solutions
>  
> 
> On Thu, Jun 27, 2013 at 7:21 PM, Ashley B <ashley at midletearth.com> wrote:
> Gerald
>  
> Were you able to resolve this issue?
> I am experiencing the exact same problem.
>  
> Rgds
> Ashley.
>  
> 
> 
> ---------- Forwarded message ----------
> From: Gerald Weber <gerald.weber at besharp.at>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Date: Thu, 27 Jun 2013 16:12:35 +0000
> Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
> I've played a little more and tried the demo at www.webrtc.org, which works to chrome should be fine.
>  
> I called another ext (snom) instead of funny prompts in the switch:
> Audio from the headset mic  -> snom is ok but not the other way round. The headset dont get any audio...
>  
> Fsctl debug is here:
> http://pastebin.freeswitch.org/21120
>  
> Pcap trace done on the fs server with pcapsipdump is here (hope this is correct):
> https://dl.dropboxusercontent.com/u/21597352/20130627-175955-2005-2002-73635e1b-59e5-1231-40bb-0050568c001c.pcap
>  
>  
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
> Gesendet: Donnerstag, 27. Juni 2013 17:27
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
>  
> try fsctl debug_level 10 and get a pcap for more info.
>  
> It looks like before it was hairpinning then figuring out the local addr and after you added localnet.auto it was right from the start, either way it should have worked.  
>  
> 
> On Thu, Jun 27, 2013 at 9:51 AM, Gerald Weber <gerald.weber at besharp.at> wrote:
> Nope, windows firewall is disabled, also iptables on FreeSWITCH Testserver is disabled.
> Every other client is working (phonerlite, bria, xlite,etc)
> 
> -----Ursprüngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Kristian Kielhofner
> Gesendet: Donnerstag, 27. Juni 2013 16:34
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
> 
> The traces you provided look fine...  Do you have a firewall running on your test PC, by chance?
> 
> On Thu, Jun 27, 2013 at 10:07 AM, Gerald Weber <gerald.weber at besharp.at> wrote:
> > Ps: i also included the output from jssip in the pastebin link
> >
> >
> >
> > Von: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
> > Anthony Minessale
> >
> >
> > Gesendet: Donnerstag, 27. Juni 2013 15:16
> > An: FreeSWITCH Users Help
> > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new
> > 1.4 BETA.
> >
> >
> >
> > try adding the sofia profile param
> >
> >
> >
> > apply-candidate-acl=localnet.auto
> >
> >
> >
> >
> >
> >
> >
> > On Thu, Jun 27, 2013 at 4:45 AM, Gerald Weber
> > <gerald.weber at besharp.at>
> > wrote:
> >
> > I got rid of sip_secure_media and tried again, here is the trace:
> >
> > http://pastebin.freeswitch.org/21116
> >
> >
> >
> > SIP User 2020 on Chrome calls Ext 20000 (the funny prompts thing)
> >
> > I'm no expert, but i guess there is something wrong with this
> > stun/run/ice thing.
> >
> >
> >
> > Thanks,
> >
> > gw
> >
> >
> >
> > Von: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
> > Anthony Minessale
> > Gesendet: Mittwoch, 26. Juni 2013 17:24
> >
> >
> > An: FreeSWITCH Users Help
> > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new
> > 1.4 BETA.
> >
> >
> >
> > You don't have to set anything in the dialplan.
> >
> > Get a debug console trace and post it to pastebin.
> >
> >
> >
> >
> >
> > On Wed, Jun 26, 2013 at 9:41 AM, Gerald Weber
> > <gerald.weber at besharp.at>
> > wrote:
> >
> > Hi,
> >
> >
> >
> > how did you get audio working ?
> >
> > I've also tried jssip and latest chrome and sip works, but there is no
> > audio. Codecs like opus,   ilbc and isac are installed.
> >
> > In the dialplan i set sip_secure_media=true but the speaker stays silent.
> > I've tried ws and wss bindings but no luck.
> >
> >
> >
> > Thanks for any suggestions,
> >
> > gw
> >
> >
> >
> > Von: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
> > João Mesquita
> > Gesendet: Mittwoch, 26. Juni 2013 15:28
> > An: FreeSWITCH Users Help
> > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new
> > 1.4 BETA.
> >
> >
> >
> > Whooray!!!! I've installed the new beta yesterday and tried it out
> > with
> > Chrome+JsSIP and it works fantastic!!!
> >
> >
> >
> > One question tho, is it possible/desirable to have a sofia profile to
> > work only with webrtc and not with udp or tcp bindings?
> >
> >
> >
> > I haven't tried and I guess I should've before asking dumb questions but....
> >
> >
> > João Mesquita
> > FreeSWITCHT Solutions
> >
> >
> >
> > On Tue, Jun 25, 2013 at 7:42 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >
> > Now also working in latest FireFox 22
> >
> >
> >
> > sipml5 works with both chrome and FF
> >
> >
> >
> > https://webrtc.freeswitch.org/sipml5/
> >
> >
> >
> >
> >
> > On Tue, Jun 25, 2013 at 2:04 AM, Michael Jerris <mike at jerris.com> wrote:
> >
> > Media is not handled by websockets.  Web-sockets are a new transport
> > for sip signaling.  If an endpoint registers from a websocket
> > transport, when we send it a call, we automatically send it an sdp
> > appropriate for webrtc.  If you want to send a call with an sdp for
> > webrtc that isn't for a registration over websockets, just add
> > media_webrtc=true var to the origination vars.  If we receive an sdp from webrtc, we will automatically handle it.
> >
> > Mike
> >
> >
> > On Jun 25, 2013, at 12:37 AM, Jeff Leung <jleung at v10networks.ca> wrote:
> >
> >> Media wise, how would that be handled with web sockets?
> >>
> >>> -----Original Message-----
> >>> From: freeswitch-users-bounces at lists.freeswitch.org
> >>> [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of
> >>> Dan Barua
> >>> Sent: Monday, June 24, 2013 3:10 PM
> >>> To: freeswitch-users at lists.freeswitch.org
> >>> Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to
> >>> new
> >>> 1.4 BETA.
> >>>
> >>> I asked the same question on Google+, Michael Jerris helpfully replied:
> >>>
> >>> you just need to setup FreeSWITCH as normal, and add 2 configuration
> >>> params for the binding addresses for the websockets and secure
> >>> websockets.  the params are ws-binding and wss-binding format is
> >>> ip:port or just :port if your using the same ip as sip-ip
> >>>
> >>>
> >>> I haven't got round to trying it myself yet.
> >>>
> >>> -Dan
> >>>
> >>>
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