[asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients

[Olli Heiskanen]

Hi,

Thanks Eric for your reply, yes I know Asterisk replaces the sdp, however
it should create ice lines when calling to a webrtc client, which it is
currently not doing.

To recap my problem (check previous messages for details); I have 2 webrtc
clients (sip.js on chrome) with realtime information that appears to be
correct. When calling from A to B, INVITE coming to Asterisk contains
correct sdp, but when the INVITE leaves Asterisk, the sdp lacks ice lines.

cheers,
Olli

2014-10-02 18:13 GMT+03:00 Eric Wieling <EWieling at nyigc.com>:

> Asterisk is not a SIP Proxy.   It is a B2BUA and will **always** replace
> the SDP with its own.
>
>
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Olli Heiskanen
> *Sent:* Thursday, October 02, 2014 9:06 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk removes ice lines in sdp when
> calling between webrtc clients
>
>
>
>
> Hi,
>
>
>
> Is there anything I can do with this problem? Re-installing Asterisk does
> not solve this and the problem still persists. Or is there any other logs
> or configurations I can provide to help figure out why Asterisk is removing
> lines from the sdp?
>
>
>
> Any ideas would be greatly appreciated! I also tried removing everything
> under /etc/asterisk/ and make samples to restore any errors I could have
> had in my configurations, then restoring my minimal configuration:
> asterisk.conf, extconfig.conf, extensions.conf, res_mysql.conf and
> sip.conf. This did not help.
>
>
>
> (in case this message comes double, I just canceled posting of previous
> similar one as it was too big)
>
>
>
> cheers,
>
> Olli
>
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