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@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@gmail.com <mailto:red.rain.seven@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 <http://stun.server.com> "? Thanks, Henry On Fri, Jun 28, 2013 at 4:45 PM, Anthony Minessale <anthony.minessale@gmail.com <mailto:anthony.minessale@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@gmail.com <mailto:red.rain.seven@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@gmail.com <mailto:anthony.minessale@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@gmail.com <mailto:red.rain.seven@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@5g4f1tqaruji.invalid] has been answered 2013-06-28 11:52:01.271125 [DEBUG] switch_channel.c:3595 (sofia/internal/sip:9ku10lcl@5g4f1tqaruji.invalid) Callstate Change RINGING -> ACTIVE 2013-06-28 11:52:01.291128 [DEBUG] switch_core_codec.c:244 sofia/internal/1003@72.1.46.122 <mailto:1003@72.1.46.122> Restore previous codec PCMU:0. 2013-06-28 11:52:01.291128 [DEBUG] mod_sofia.c:822 Local SDP sofia/internal/1003@72.1.46.122 <mailto:1003@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@besharp.at <mailto:gerald.weber@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@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@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@midletearth.com <mailto:ashley@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@besharp.at <mailto:gerald.weber@besharp.at> > To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@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 <http://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/21597352/20130627-175955-2005-2002-73635... Von: freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@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@besharp.at <mailto:gerald.weber@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@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@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@besharp.at <mailto:gerald.weber@besharp.at> > wrote:
Ps: i also included the output from jssip in the pastebin link
Von: freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@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@besharp.at <mailto:gerald.weber@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@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@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@besharp.at <mailto:gerald.weber@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@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@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@gmail.com <mailto:anthony.minessale@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@jerris.com <mailto:mike@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@v10networks.ca <mailto:jleung@v10networks.ca> > wrote:
Media wise, how would that be handled with web sockets?
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch- <mailto:freeswitch-> users-bounces@lists.freeswitch.org <mailto:users-bounces@lists.freeswitch.org> ] On Behalf Of Dan Barua Sent: Monday, June 24, 2013 3:10 PM To: freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@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
-- View this message in context: http://freeswitch- users.2379917.n2.nabble.com/FreeSWITCH-adds-WebRTC-support-to-new- <http://users.2379917.n2.nabble.com/FreeSWITCH-adds-WebRTC-support-to-new-> 1-4-BETA-tp7592138p7592141.html Sent from the freeswitch-users mailing list archive at Nabble.com.
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___ Professional FreeSWITCH Consulting Services:
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<tel:%2B19193869900> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 <tel:%2B19193869900> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 <tel:%2B19193869900> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org