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

[Joel Serrano]

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 at 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 at gmail.com> wrote:
>
>> Hello guys,
>>
>> Any help on this?
>>
>> Thanks
>>
>> David
>> ᐧ
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>
>> On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil <
>> david.villasmil.work at 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 at gmail.com
>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>> ᐧ
>>>
>>
>>
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>
>
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