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

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

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
ᐧ


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_________________________________________________________________________
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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
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