[Freeswitch-users] sip to webrtc - sdp invalid description

[David Villasmil]

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 at gmail.com
phone: +34669448337 <+34%20669%2044%2083%2037>
ᐧ
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