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

MARAND, Remi rmarand at prosodie.com
Wed Dec 12 17:42:10 UTC 2018


Sorry, i come back with this case.

I understand that the browsers behaviour is to use DTLS... but in my case it is not the case.

I have to set this constraints in my client javascript, and i do not find how to code it with sip.js....

Should you please give me a way to set correctly that on the Freeswitch side, a configuration option to specify « WebRTC Media » on connection ??

I tried with <action application="set" data="sip_secure_media=false"/> in the dialplan/default.xml file, but it does not change, i tried before with {media_webrtc=true} in the extension file without success.

Thanks for help.

Rémi Marand.


De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> De la part de Jeremy Lainé
Envoyé : lundi 19 novembre 2018 09:18
À : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC

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:


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

On 11/13/18 8:55 AM, MARAND, Remi wrote:

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.


Remi Marand.
rmarand at prosodie.com<mailto:rmarand at prosodie.com>

De : MARAND, Remi
Envoyé : lundi 5 novembre 2018 18:01
À : freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Objet : One way audio case between FS and Chrome on WebRTC


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 :
@IP for Chrome :

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.


Rémi Marand - Product Owner - Pod Connect.
PROSODIE - Marketing & Produit
Tél. : +33 (0) /
rmarand at prosodie.com<mailto:rmarand at prosodie.com>

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