[Freeswitch-users] One way audio case between FS and Chrome on WebRTC

Jeremy Lainé jeremy.laine at m4x.org
Mon Nov 19 08:17:51 UTC 2018


That is very odd, the default behaviour in browsers *is* to use DTLS for
SRTP keying instead of the legacy SDES. In fact SDES's status is "MUST
NOT implement" since 2013, and the DtlsSrtpKeyAgreement constraint is
scheduled for retirement:

https://bugs.chromium.org/p/chromium/issues/detail?id=804275

Concerning SIP.js I have never experienced the need to set this
constraint, it works fine against freeswitch without it.

Jeremy

On 11/13/18 8:55 AM, MARAND, Remi wrote:
>
> Hello,
>
>  
>
> I finally was able to solve my problem.
>
>  
>
> For information, on this trouble, the parameter :
>
> /RTCConstraints: {"optional": [{'DtlsSrtpKeyAgreement': 'true'}]}/
>
> Is to add at the good place in jquery.FSRTC.js (or verto-min.js (I
> think that's already the case in this script)) i did not find how to
> code in the sip.js version but it should be possible.
>
>  
>
> Thanks to those who answered my question, and sorry for the 3 Mb of
> pcap file i sent to the user-list !!!
>
>  
>
> Perhaps this DtlsSrtpKeyAgreement parameter role should be added and
> explain in the Verto/WebRTC examples availables on Websites, i suppose
> that in 2014, it was not mandatory but now with the lasts versions of
> Chrome and FF it seems to be.
>
>  
>
> Regards.
>
>  
>
> Remi Marand.
>
> rmarand at prosodie.com <mailto:rmarand at prosodie.com>
>
> +33687725325.
>
>  
>
> *De :*MARAND, Remi
> *Envoyé :* lundi 5 novembre 2018 18:01
> *À :* freeswitch-users at lists.freeswitch.org
> *Objet :* One way audio case between FS and Chrome on WebRTC
>
>  
>
> Hello,
>
>  
>
> I am trying to validate FS as a SIP to WebRTC Gateway in our lab
> environment.
>
> I started in middle October and have good result on it, but i cannot
> understand this One Way Audio trouble.
>
>  
>
> I must thank the Freeswitch team and contributors for this very
> impressive work.
>
>  
>
> FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit)
>
> On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux
>
> Openssl version : OpenSSL 1.1.0f  25 May 2017
>
> Chrome version: 69.0 (I tried with different version and with Firefox
> with the same trouble).
>
>  
>
> The wss part is ok with sip.js and verto.js
>
>  
>
> The Ice negotiation is ok, I use sometimes local networks and
> sometimes web, I have had to authorize networks in the candidate ACL
> and domain ACL (acl.conf.xml) The result is the same on both topology.
>
>  
>
> DTLS negotiation is OK, and there is UDP streams between Chrome (or
> Firefox) and FS in both ways.
>
>  
>
> There is no “audible” audio in the direction from FS to Chrome, the
> other direction is OK.
>
>  
>
> The simplest test is to call the 5000 number from the Chrome client, I
> send you a paste bin and pcap trace for this call.
>
>  
>
> Should you give me information element to progress on this, what is
> really mandatory in the sip_profile/internal.xml and external.xml
> files, and in directory/default/1000.xml for a WebRTC call ?? What
> should be the good options in fs_cli to see if the encryption of RTP
> packets is ok or not.. ?
>
> Do you think that I have to reinstall a Freeswitch from the current
> branch ?
>
>  
>
> @IP for FS : 192.168.145.67
>
> @IP for Chrome : 10.70.54.43
>
>  
>
> Link on the pastebin : https://pastebin.freeswitch.org/view/09a72087
>
>  
>
> I have a pcap on the same call that I can provide (3 Mb) if necessary..
>
>  
>
> Thank you for helping me !!
>
>  
>
> Best regards.
>
>  
>
> Prosodie-signature <http://www.prosodie.com/>
>
> 	
>
> *Rémi Marand – Product Owner – Pod Connect.
> *PROSODIE – Marketing & Produit
> Tél. : +33 (0)1.46.84.12.77 / 06.87.72.53.25
> rmarand at prosodie.com <mailto:rmarand at prosodie.com>
>
>  
>
>  
>
>  
>
>  
>
>  
>
>  
>
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