Can't call from Firefox 22 to Freeswitch using sipml5
Hi, I want to make calls using sipml5 and freeswitch. I have installed freeswitch from the git repository in an ec2 instance with elastic ip address, configured sip_profiles/internal.xml with the EIP and here the results: Using Google Chrome and the sipml5 client I register as user 1001 with password 1234 , I dial 5000 and I hear the IVR voice, also I can join to a conference I created, and hear myself in an echo test dialing 9196. No problem with that. Using Firefox 22 and the same sipml5 client I register as user 1002 and password 1234, then dial 5000, I see the audio files are played in the freeswitch cli but I can not hear anything. I guess the problem is not in my sipml5 client neither Firefox because when I use them to register to webrtc.freeswitch.org there is not any problem, I hear the voice after dialing 5000. I think I am missing some configuration in my Freeswitch server, where I also loaded the opus module and set OPUS in the file vars.xml , in the parameters global_codec_prefs and outbound_codec_prefs. I attach the console output from Freeswitch and the console output from Firefox Here the freeswitch output http://pastebin.com/S0yJxT1J Here the Firefox console output http://pastebin.com/8kBGdY3J Thanks in advance for the help -- Gustavo Salazar
Hi all, i have still the same problem, with 1.4beta branch and master branch. Firefox 22 still no audio. Did you get any solution with your Problem? 2013/7/26 Gustavo Salazar <guga.salazar.loor@gmail.com>
Hi,
I want to make calls using sipml5 and freeswitch. I have installed freeswitch from the git repository in an ec2 instance with elastic ip address, configured sip_profiles/internal.xml with the EIP and here the results:
Using Google Chrome and the sipml5 client I register as user 1001 with password 1234 , I dial 5000 and I hear the IVR voice, also I can join to a conference I created, and hear myself in an echo test dialing 9196. No problem with that.
Using Firefox 22 and the same sipml5 client I register as user 1002 and password 1234, then dial 5000, I see the audio files are played in the freeswitch cli but I can not hear anything.
I guess the problem is not in my sipml5 client neither Firefox because when I use them to register to webrtc.freeswitch.org there is not any problem, I hear the voice after dialing 5000.
I think I am missing some configuration in my Freeswitch server, where I also loaded the opus module and set OPUS in the file vars.xml , in the parameters global_codec_prefs and outbound_codec_prefs.
I attach the console output from Freeswitch and the console output from Firefox
Here the freeswitch output http://pastebin.com/S0yJxT1J
Here the Firefox console output http://pastebin.com/8kBGdY3J
Thanks in advance for the help
-- Gustavo Salazar
-- Mit freundlichen Grüßen *Karsten Horsmann*
Hi Karsten, I haven't solved the problem, it works only with Google Chrome. I tested the master branch and the version 1.4 with the same results. .. 2013/8/5 Karsten Horsmann <khorsmann@gmail.com>
Hi all,
i have still the same problem, with 1.4beta branch and master branch. Firefox 22 still no audio. Did you get any solution with your Problem?
2013/7/26 Gustavo Salazar <guga.salazar.loor@gmail.com>
Hi,
I want to make calls using sipml5 and freeswitch. I have installed freeswitch from the git repository in an ec2 instance with elastic ip address, configured sip_profiles/internal.xml with the EIP and here the results:
Using Google Chrome and the sipml5 client I register as user 1001 with password 1234 , I dial 5000 and I hear the IVR voice, also I can join to a conference I created, and hear myself in an echo test dialing 9196. No problem with that.
Using Firefox 22 and the same sipml5 client I register as user 1002 and password 1234, then dial 5000, I see the audio files are played in the freeswitch cli but I can not hear anything.
I guess the problem is not in my sipml5 client neither Firefox because when I use them to register to webrtc.freeswitch.org there is not any problem, I hear the voice after dialing 5000.
I think I am missing some configuration in my Freeswitch server, where I also loaded the opus module and set OPUS in the file vars.xml , in the parameters global_codec_prefs and outbound_codec_prefs.
I attach the console output from Freeswitch and the console output from Firefox
Here the freeswitch output http://pastebin.com/S0yJxT1J
Here the Firefox console output http://pastebin.com/8kBGdY3J
Thanks in advance for the help
-- Gustavo Salazar
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Gustavo Salazar
Hi, Try Firefox 23 (nightly build). It should have better webrtc support. Regards On 05/08/2013 16:16, Gustavo Salazar wrote:
Hi Karsten,
I haven't solved the problem, it works only with Google Chrome. I tested the master branch and the version 1.4 with the same results. ..
2013/8/5 Karsten Horsmann <khorsmann@gmail.com <mailto:khorsmann@gmail.com>>
Hi all,
i have still the same problem, with 1.4beta branch and master branch. Firefox 22 still no audio. Did you get any solution with your Problem?
2013/7/26 Gustavo Salazar <guga.salazar.loor@gmail.com <mailto:guga.salazar.loor@gmail.com>>
Hi,
I want to make calls using sipml5 and freeswitch. I have installed freeswitch from the git repository in an ec2 instance with elastic ip address, configured sip_profiles/internal.xml with the EIP and here the results:
Using Google Chrome and the sipml5 client I register as user 1001 with password 1234 , I dial 5000 and I hear the IVR voice, also I can join to a conference I created, and hear myself in an echo test dialing 9196. No problem with that.
Using Firefox 22 and the same sipml5 client I register as user 1002 and password 1234, then dial 5000, I see the audio files are played in the freeswitch cli but I can not hear anything.
I guess the problem is not in my sipml5 client neither Firefox because when I use them to register to webrtc.freeswitch.org <http://webrtc.freeswitch.org> there is not any problem, I hear the voice after dialing 5000.
I think I am missing some configuration in my Freeswitch server, where I also loaded the opus module and set OPUS in the file vars.xml , in the parameters global_codec_prefs and outbound_codec_prefs.
I attach the console output from Freeswitch and the console output from Firefox
Here the freeswitch output http://pastebin.com/S0yJxT1J
Here the Firefox console output http://pastebin.com/8kBGdY3J
Thanks in advance for the help
-- Gustavo Salazar
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ 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
-- Gustavo Salazar
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Cezary, but the question is, why Firefox 22 works with sipml5 + https://webrtc.freeswitch.org/sipml5/ and *not* with my installation? 2013/8/5 Cezary Siwek <csiwek@orange-vallee.net>
Hi,
Try Firefox 23 (nightly build). It should have better webrtc support.
Regards
-- Mit freundlichen Grüßen *Karsten Horsmann*
The ice not ready message comes from not getting any stun messages to the candidate ip/port sent in the 200ok. Maybe do a pcap and see if the firefox is sending any stun packets to that addr. On Mon, Aug 5, 2013 at 1:19 PM, Karsten Horsmann <khorsmann@gmail.com>wrote:
Hi Cezary,
but the question is, why Firefox 22 works with sipml5 + https://webrtc.freeswitch.org/sipml5/ and *not* with my installation?
2013/8/5 Cezary Siwek <csiwek@orange-vallee.net>
Hi,
Try Firefox 23 (nightly build). It should have better webrtc support.
Regards
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Hello Anthony, i made pcaps from the FS-Server and a Firefox 22 Client. Filtered on STUN and ICMP. https://webrtc.0x300.de/pcap/client-stun.pcap 31K https://webrtc.0x300.de/pcap/server-stun.pcap 8K It seems that my gateway sends icmp (port unreachable), but i dunno why - whats so diffrent between the Chrome doing and Firefox? Regards 2013/8/5 Anthony Minessale <anthony.minessale@gmail.com>
The ice not ready message comes from not getting any stun messages to the candidate ip/port sent in the 200ok. Maybe do a pcap and see if the firefox is sending any stun packets to that addr.
-- Mit freundlichen Grüßen *Karsten Horsmann*
Make sure its going to the ports that are advertised in the sdp and that there are no nat or firewall issues in between. On Mon, Aug 5, 2013 at 2:58 PM, Karsten Horsmann <khorsmann@gmail.com>wrote:
Hello Anthony,
i made pcaps from the FS-Server and a Firefox 22 Client. Filtered on STUN and ICMP.
https://webrtc.0x300.de/pcap/client-stun.pcap 31K https://webrtc.0x300.de/pcap/server-stun.pcap 8K
It seems that my gateway sends icmp (port unreachable), but i dunno why - whats so diffrent between the Chrome doing and Firefox?
Regards
2013/8/5 Anthony Minessale <anthony.minessale@gmail.com>
The ice not ready message comes from not getting any stun messages to the candidate ip/port sent in the 200ok. Maybe do a pcap and see if the firefox is sending any stun packets to that addr.
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Hi Anthony, how is the best way to check that? With wireshark most stuff is ssl unlike normal UDP/SIP stuff. Iam still confused, that it works with webrtc.freeswitch.org at work and at home, but not with my example setup. I checked for iptables (everything fine), and its no cloud virtual foo where the server is running. Thanks in advance, Regards 2013/8/5 Anthony Minessale <anthony.minessale@gmail.com>
Make sure its going to the ports that are advertised in the sdp and that there are no nat or firewall issues in between.
-- Mit freundlichen Grüßen *Karsten Horsmann*
Your example might have different network conditions than the demo. You can check by enabling sip trace on FS and bring up the call and look at the ports that are advertised out. You can try turning off selinux or iptables as a test. On Mon, Aug 5, 2013 at 3:41 PM, Karsten Horsmann <khorsmann@gmail.com>wrote:
Hi Anthony,
how is the best way to check that? With wireshark most stuff is ssl unlike normal UDP/SIP stuff.
Iam still confused, that it works with webrtc.freeswitch.org at work and at home, but not with my example setup.
I checked for iptables (everything fine), and its no cloud virtual foo where the server is running.
Thanks in advance, Regards
2013/8/5 Anthony Minessale <anthony.minessale@gmail.com>
Make sure its going to the ports that are advertised in the sdp and that there are no nat or firewall issues in between.
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Karsten, I executed the openssl commands against my local server and webrtc.freeswitch.org, both show the same output that you described against a broken host. Using Firefox I still have the same results: Firefox -> 5000 -> webrtc.freeswitch.org ....... GOOD Firefox -> 5000 -> my local vm .......................NO AUDIO And now using Chrome: Chrome -> 5000 -> webrtc.freeswitch.org ....... NO AUDIO This worked before Chrome -> -> 5000 -> my local vm ......................GOOD I am using Chrome 29.0.1547.57 m Karsten, did the suggestions from Anthony work for you? Anthony, In the freeswitch cli I see this message many times 2013-08-22 11:42:55.504980 [ALERT] switch_rtp.c:5662 Skip sending audio packet 98 bytes (dtls not ready!) Do you have something related to dtls configured in webrtc.freeswitch.org? 2013/8/5 Anthony Minessale <anthony.minessale@gmail.com>
Your example might have different network conditions than the demo. You can check by enabling sip trace on FS and bring up the call and look at the ports that are advertised out. You can try turning off selinux or iptables as a test.
On Mon, Aug 5, 2013 at 3:41 PM, Karsten Horsmann <khorsmann@gmail.com>wrote:
Hi Anthony,
how is the best way to check that? With wireshark most stuff is ssl unlike normal UDP/SIP stuff.
Iam still confused, that it works with webrtc.freeswitch.org at work and at home, but not with my example setup.
I checked for iptables (everything fine), and its no cloud virtual foo where the server is running.
Thanks in advance, Regards
2013/8/5 Anthony Minessale <anthony.minessale@gmail.com>
Make sure its going to the ports that are advertised in the sdp and that there are no nat or firewall issues in between.
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Gustavo Salazar
Hi Gustavo, i think there am more than one Problem with WebRTC + sipml5 + Firefox 22 (or higher). First, i saw an dtls error and many "use 1.0.1e openssl" hints. You can check this with openssl client. Here a good and a broken example: openssl s_client -dtls1 -connect webrtc.freeswitch.org:7443 CONNECTED(00000003) openssl s_client -dtls1 -connect some.broken.host:7443 CONNECTED(00000003) write:errno=111 no peer certificate available No client certificate CA names sent SSL handshake has read 0 bytes and written 0 bytes New, (NONE), Cipher is (NONE) Secure Renegotiation IS NOT supported Compression: NONE Expansion: NONE SSL-Session: Protocol : DTLSv1 Cipher : 0000 Session-ID: Session-ID-ctx: Master-Key: Key-Arg : None Krb5 Principal: None PSK identity: None PSK identity hint: None Start Time: 1375716502 Timeout : 7200 (sec) Verify return code: 0 (ok) Then you can check https://webrtc.freeswitch.org/sipml5/ with Firefox 22. It works like a charm. I thought then, maybe some voodoo was made on the sipml5 code. But there was a few settings insert. And i test exact the same sipml5 code with my test WebRTC FreeSWITCHes, it doesnt work with Firefox. Only Chrome works. In Firefox Webdeveloper Console i got only an SDP Warning of the sipml5 Client: ({name:"INVALID_SESSION_DESCRIPTION", message:"Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning: Unrecognized attribute (msid-semantic) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) ", __exposedProps__:{name:"rw", message:"rw"}}) Maybe the developers give us another hint? Cheers... 2013/8/5 Gustavo Salazar <guga.salazar.loor@gmail.com>
Hi Karsten,
I haven't solved the problem, it works only with Google Chrome. I tested the master branch and the version 1.4 with the same results. ..
-- Mit freundlichen Grüßen *Karsten Horsmann*
try going to your wss address in firefox as an https addr. Replace wss:// with https:// and paste it in the location toolbar. You may get the trust warning and have to set exception on the cert. They won't tell you this when you hit it as wss and it will not work silently forever. Once you trust the cert over https it will allow it for subsequent hits as wss. On Mon, Aug 5, 2013 at 11:59 AM, Karsten Horsmann <khorsmann@gmail.com>wrote:
Hi Gustavo,
i think there am more than one Problem with WebRTC + sipml5 + Firefox 22 (or higher).
First, i saw an dtls error and many "use 1.0.1e openssl" hints.
You can check this with openssl client. Here a good and a broken example:
openssl s_client -dtls1 -connect webrtc.freeswitch.org:7443 CONNECTED(00000003)
openssl s_client -dtls1 -connect some.broken.host:7443 CONNECTED(00000003) write:errno=111 no peer certificate available No client certificate CA names sent SSL handshake has read 0 bytes and written 0 bytes New, (NONE), Cipher is (NONE) Secure Renegotiation IS NOT supported Compression: NONE Expansion: NONE SSL-Session: Protocol : DTLSv1 Cipher : 0000 Session-ID: Session-ID-ctx: Master-Key: Key-Arg : None Krb5 Principal: None PSK identity: None PSK identity hint: None Start Time: 1375716502 Timeout : 7200 (sec) Verify return code: 0 (ok)
Then you can check https://webrtc.freeswitch.org/sipml5/ with Firefox 22. It works like a charm.
I thought then, maybe some voodoo was made on the sipml5 code. But there was a few settings insert. And i test exact the same sipml5 code with my test WebRTC FreeSWITCHes, it doesnt work with Firefox.
Only Chrome works.
In Firefox Webdeveloper Console i got only an SDP Warning of the sipml5 Client:
({name:"INVALID_SESSION_DESCRIPTION", message:"Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning: Unrecognized attribute (msid-semantic) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) ", __exposedProps__:{name:"rw", message:"rw"}})
Maybe the developers give us another hint?
Cheers...
2013/8/5 Gustavo Salazar <guga.salazar.loor@gmail.com>
Hi Karsten,
I haven't solved the problem, it works only with Google Chrome. I tested the master branch and the version 1.4 with the same results. ..
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Hello Anthony, i did that and Firefox dont said its an untrused certificate. Its an trusted certificate. I made an debug-Logfile and test it with git-master. d1268e81036ce8ce00de8ee22f387cdbf43a7203 switch_rtp.c:5668 Skip sending audio packet 88 bytes (ice not ready!) is to see in the logfile over and over again, if the rtp should send. FreeSWITCH Debug-Logfile: *http://pastebin.freeswitch.org/21270* [19:17:03.572] ({name:"INVALID_SESSION_DESCRIPTION", message:"Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning: Unrecognized attribute (msid-semantic) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) | SDP Parsing Error: Warning: Unrecognized attribute (ssrc) ", __exposedProps__:{name:"rw", message:"rw"}}) Firefox 22 Webdeveloper-Logfile: *http://pastebin.freeswitch.org/21268* Hope everything is in there to take a look. Cheers. Karsten 2013/8/5 Anthony Minessale <anthony.minessale@gmail.com>
try going to your wss address in firefox as an https addr. Replace wss:// with https:// and paste it in the location toolbar.
You may get the trust warning and have to set exception on the cert. They won't tell you this when you hit it as wss and it will not work silently forever. Once you trust the cert over https it will allow it for subsequent hits as wss.
-- 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Mit freundlichen Grüßen *Karsten Horsmann*
participants (4)
-
Anthony Minessale
-
Cezary Siwek
-
Gustavo Salazar
-
Karsten Horsmann