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

Joel Serrano joel at gogii.net
Sun Jul 9 02:56:59 UTC 2017


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