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

[MARAND, Remi]

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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