[Freeswitch-users] Freeswitch sends 2 SDP answers which causes webrtc client to drop the call

[Georgi Nikolaev]

Hi Michael,

Thanks for looking into it.

FS Log: http://pastebin.com/LzYAdzQG
Chrome log: http://pastebin.com/jcL1QkyJ
Webrtc-internals: http://pastebin.com/YNkK2dgV

By 2 SDP answers I mean I receive 1 with the 183 SIP message I right after
I get an 200 OK with SDP again which confuses the webrtc client. According
to the chaps from the webrtc group and the sipjs group, this is not allowed
in the webrtc standard. In my original email you can find the links to the
discussions I had with them as well.

Let me know if anything else is needed.

Thank you!

On Tue, Feb 14, 2017 at 4:50 PM, Michael Jerris <mike at jerris.com> wrote:

> not sure what 2 sdp answers mean.  Do you have a freeswitch debug log with
> sip trace enabled?
>
> On Feb 14, 2017, at 9:36 AM, Georgi Nikolaev <georgi at proitc.org> wrote:
>
> Hi Folks,
>
> I am experiencing an issue where remote SDP can not be set due to
> receiving 2 SDP answers.
> The setup is webrtc (sipjs)->Freeswitch->SIP Termination, where a call is
> originated by the sipjs.ua calling a public number. As soon as the remote
> party answers the call, the call drops due to not being able to gather the
> remote media path.
>
> Worth mentioning is that only some calls fail where the majority are
> successful. The difference between the successful and the unsuccessful ones
> is in the SDP messages. The good one receive ‘answer' for UDP/TLS/RTP/SAVPF
> followed by ‘offer’ for RTP/SAVPF.
> The one that fail are 2 SDP answers.
>
> I have discussed the topic on the webrtc group: https://groups.google.
> com/forum/?utm_medium=email&utm_source=footer#!msg/
> discuss-webrtc/IwOxU-IcxdY/Hfn02tEfCwAJ
> Also raised this in the sipjs forum: https://groups.google.
> com/forum/?utm_medium=email&utm_source=footer#!topic/sip_js/ABy98d3Kwu8
>
> On both it has been suggested that this is either a setting in Freeswitch
> or its incorrectly sent SDP messages.
>
> Freeswitch version: 1.6.14
> SIP.js: 0.7.5
> Browser: Chrome - latest version
>
> I would appreciate any help on this as its causing me some random call
> drops affecting the service.
>
> Here are the logs as well:
> FS log: http://pastebin.com/UQW6AdcQ
> Chrome log: http://pastebin.com/Crk69MrS
> webrtc-internals-log: http://pastebin.com/BVr7v4EB
>
> Any ideas/suggestions/comments are welcome.
>
> Thank you in advance.
>
> Best Regards,
> Georgi Nikolaev
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