sip to webrtc - sdp invalid description
Hello guys, I have this setup: Zoiper-->Kamailio->fs->kamailio->webrtc client(s) Whenever webrtc clients call each other, calls are ok. But when the zoiper (regular sip/tcp) calls, the browsers complaint about: "no ice-ufrag" (firefox) or "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." (Chrome). I am setting the sdp in freeswitch as: <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <action application="export" data="zrtp_secure_media=false"/> <action application="export" data="nolocal:rtp_secure_media=true"/> and the actual sdp is: v=0 o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 40954 RTP/SAVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVrVfjA0Im Wgu5WyLK0vtg60RTk6Koe8c0sRkjU a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw06I0Xok/9KbRystQ a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ ETPuCOiOIxjFLRjbFbDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAMpmok06k W9ZGDgH3/Lc3ZytVn2SR4 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixRhwYdiIh E6r3u5CCVA a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqNDAIiJrb OqOKb+erlhPQtBKF4wzomjbN0sBIiE4w a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzdvtfnw6A ARU3+jGA0WzTSDlDd20vI a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8sNFzuZzi Ng64KuvfNS a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3dhTqD0X W8FAOLNtmP a=ptime:20 a=nortpproxy:yes I'm pretty new to the webrtc stuff, so any help is greatly appreciated! Thanks! David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ
Hello guys, Any help on this? Thanks David ᐧ Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
I have this setup:
Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
Whenever webrtc clients call each other, calls are ok. But when the zoiper (regular sip/tcp) calls, the browsers complaint about:
"no ice-ufrag" (firefox) or "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." (Chrome).
I am setting the sdp in freeswitch as:
<action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <action application="export" data="zrtp_secure_media=false"/> <action application="export" data="nolocal:rtp_secure_media=true"/>
and the actual sdp is:
v=0 o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 40954 RTP/SAVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVr VfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw0 6I0Xok/9KbRystQ a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ETPuCOiOIxjFLRjbF bDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAM pmok06kW9ZGDgH3/Lc3ZytVn2SR4 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixR hwYdiIhE6r3u5CCVA a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqN DAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzd vtfnw6AARU3+jGA0WzTSDlDd20vI a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8s NFzuZziNg64KuvfNS a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3 dhTqD0XW8FAOLNtmP a=ptime:20 a=nortpproxy:yes
I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
Thanks!
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ
Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS? WebRTC requires SRTP-DTLS. On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
Any help on this?
Thanks
David ᐧ
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037>
On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
I have this setup:
Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
Whenever webrtc clients call each other, calls are ok. But when the zoiper (regular sip/tcp) calls, the browsers complaint about:
"no ice-ufrag" (firefox) or "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." (Chrome).
I am setting the sdp in freeswitch as:
<action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <action application="export" data="zrtp_secure_media=false"/> <action application="export" data="nolocal:rtp_secure_media=true"/>
and the actual sdp is:
v=0 o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 40954 RTP/SAVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVr VfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw0 6I0Xok/9KbRystQ a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ETPuCOiOIxjFLRjbF bDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAM pmok06kW9ZGDgH3/Lc3ZytVn2SR4 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixR hwYdiIhE6r3u5CCVA a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqN DAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzd vtfnw6AARU3+jGA0WzTSDlDd20vI a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8s NFzuZziNg64KuvfNS a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3 dhTqD0XW8FAOLNtmP a=ptime:20 a=nortpproxy:yes
I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
Thanks!
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Nevermind, I missed the part that FS is in the middle. I'm going to test that and see what I find. On Sat, Jul 8, 2017 at 19:56 Joel Serrano <joel@gogii.net> wrote:
Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS?
WebRTC requires SRTP-DTLS.
On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
Any help on this?
Thanks
David ᐧ
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037>
On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
I have this setup:
Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
Whenever webrtc clients call each other, calls are ok. But when the zoiper (regular sip/tcp) calls, the browsers complaint about:
"no ice-ufrag" (firefox) or "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." (Chrome).
I am setting the sdp in freeswitch as:
<action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <action application="export" data="zrtp_secure_media=false"/> <action application="export" data="nolocal:rtp_secure_media=true"/>
and the actual sdp is:
v=0 o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 40954 RTP/SAVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVrVfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw06I0Xok/9KbRystQ a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ETPuCOiOIxjFLRjbFbDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAMpmok06kW9ZGDgH3/Lc3ZytVn2SR4 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixRhwYdiIhE6r3u5CCVA a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqNDAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzdvtfnw6AARU3+jGA0WzTSDlDd20vI a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8sNFzuZziNg64KuvfNS a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3dhTqD0XW8FAOLNtmP a=ptime:20 a=nortpproxy:yes
I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
Thanks!
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
many thanks! ᐧ Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 On Sun, Jul 9, 2017 at 7:35 AM, Joel Serrano <joel@gogii.net> wrote:
Nevermind, I missed the part that FS is in the middle. I'm going to test that and see what I find.
On Sat, Jul 8, 2017 at 19:56 Joel Serrano <joel@gogii.net> wrote:
Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS?
WebRTC requires SRTP-DTLS.
On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
Any help on this?
Thanks
David ᐧ
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037>
On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
I have this setup:
Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
Whenever webrtc clients call each other, calls are ok. But when the zoiper (regular sip/tcp) calls, the browsers complaint about:
"no ice-ufrag" (firefox) or "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." (Chrome).
I am setting the sdp in freeswitch as:
<action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <action application="export" data="zrtp_secure_media=false"/> <action application="export" data="nolocal:rtp_secure_ media=true"/>
and the actual sdp is:
v=0 o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 40954 RTP/SAVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVrVfjA0Im Wgu5WyLK0vtg60RTk6Koe8c0sRkjU a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+ D5NVoKFFw06I0Xok/9KbRystQ a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ ETPuCOiOIxjFLRjbFbDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline: evU8MzAtiSHwKb95s4V9IAMpmok06kW9ZGDgH3/Lc3ZytVn2SR4 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline: KKDkT0DssohSeKFsX6tbixRhwYdiIhE6r3u5CCVA a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline: D3JgGuOlIxXfHGdqf7lKWqNDAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline: c12ebWWzZ1cqZN0v5C5uYzdvtfnw6AARU3+jGA0WzTSDlDd20vI a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline: nPZku7S6hlz2OPAff8T9I8sNFzuZziNg64KuvfNS a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3dhTqD0X W8FAOLNtmP a=ptime:20 a=nortpproxy:yes
I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
Thanks!
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 David, I have: [iOS/Android app with Linphone SIP stack] <-> Kamailio <-> FS <-> Kamailio <-> [WebRTC] And it works correctly in all combinations (both Kamailios are only signalling, rtp goes from clients to FS directly). Tomorrow i'll send you the combination of codec parameters we are using so you can compare, we can do more tests from there with the traces. Joel. On Tue, Jul 11, 2017 at 4:58 AM, David Villasmil < david.villasmil.work@gmail.com> wrote:
many thanks! ᐧ
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037>
On Sun, Jul 9, 2017 at 7:35 AM, Joel Serrano <joel@gogii.net> wrote:
Nevermind, I missed the part that FS is in the middle. I'm going to test that and see what I find.
On Sat, Jul 8, 2017 at 19:56 Joel Serrano <joel@gogii.net> wrote:
Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS?
WebRTC requires SRTP-DTLS.
On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
Any help on this?
Thanks
David ᐧ
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037>
On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
I have this setup:
Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
Whenever webrtc clients call each other, calls are ok. But when the zoiper (regular sip/tcp) calls, the browsers complaint about:
"no ice-ufrag" (firefox) or "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." (Chrome).
I am setting the sdp in freeswitch as:
<action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <action application="export" data="zrtp_secure_media=false"/> <action application="export" data="nolocal:rtp_secure_media =true"/>
and the actual sdp is:
v=0 o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 40954 RTP/SAVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVr VfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw0 6I0Xok/9KbRystQ a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ETPuCOiOIxjFLRjbF bDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAM pmok06kW9ZGDgH3/Lc3ZytVn2SR4 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixR hwYdiIhE6r3u5CCVA a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqN DAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzd vtfnw6AARU3+jGA0WzTSDlDd20vI a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8s NFzuZziNg64KuvfNS a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3 dhTqD0XW8FAOLNtmP a=ptime:20 a=nortpproxy:yes
I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
Thanks!
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
That would be amazing, thanks! And if you could share you kamailio config, that would be awesome! On Thu, Jul 13, 2017 at 5:26 AM Joel Serrano <joel@gogii.net> wrote:
Hi David,
I have:
[iOS/Android app with Linphone SIP stack] <-> Kamailio <-> FS <-> Kamailio <-> [WebRTC]
And it works correctly in all combinations (both Kamailios are only signalling, rtp goes from clients to FS directly).
Tomorrow i'll send you the combination of codec parameters we are using so you can compare, we can do more tests from there with the traces.
Joel.
On Tue, Jul 11, 2017 at 4:58 AM, David Villasmil < david.villasmil.work@gmail.com> wrote:
many thanks! ᐧ
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037>
On Sun, Jul 9, 2017 at 7:35 AM, Joel Serrano <joel@gogii.net> wrote:
Nevermind, I missed the part that FS is in the middle. I'm going to test that and see what I find.
On Sat, Jul 8, 2017 at 19:56 Joel Serrano <joel@gogii.net> wrote:
Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS?
WebRTC requires SRTP-DTLS.
On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
Any help on this?
Thanks
David ᐧ
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037>
On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil < david.villasmil.work@gmail.com> wrote:
Hello guys,
I have this setup:
Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
Whenever webrtc clients call each other, calls are ok. But when the zoiper (regular sip/tcp) calls, the browsers complaint about:
"no ice-ufrag" (firefox) or "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." (Chrome).
I am setting the sdp in freeswitch as:
<action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <action application="export" data="zrtp_secure_media=false"/> <action application="export" data="nolocal:rtp_secure_media=true"/>
and the actual sdp is:
v=0 o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 40954 RTP/SAVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVrVfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw06I0Xok/9KbRystQ a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ETPuCOiOIxjFLRjbFbDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAMpmok06kW9ZGDgH3/Lc3ZytVn2SR4 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixRhwYdiIhE6r3u5CCVA a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqNDAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzdvtfnw6AARU3+jGA0WzTSDlDd20vI a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8sNFzuZziNg64KuvfNS a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3dhTqD0XW8FAOLNtmP a=ptime:20 a=nortpproxy:yes
I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
Thanks!
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
participants (2)
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David Villasmil
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Joel Serrano