[asterisk-dev] Asterisk 11; WEBRTC firefox nightly build fingeprint sha-256

[Daniel Pocock]

On 24/01/14 10:59, Lorenzo Miniero wrote:
> Hi Daniel,
>
> the "sha-2" error can be easily circumvented, and the dtlsverify=no
> needs an additional callback in the code to always return a success.
> Nitesh and I provided some patches here:
>
> https://issues.asterisk.org/jira/browse/ASTERISK-22961
>
> Mine was specifically targeted at getting Firefox to work, but I only
> tested incoming calls. I didn't test Nitesh's one, but apparently he
> managed to get it to work as well.
>

Thanks for this, I've tested with it

Two things were necessary for success with Firefox:
a) I applied Nitish's patch to the latest 11.7 from Debian (it is on a
branch dtls-srtp-patch), it builds on wheezy and appears to work
http://anonscm.debian.org/gitweb/?p=pkg-voip/asterisk.git;a=shortlog;h=refs/heads/dtls-srtp-patch
Anybody wanting to test can clone from there and then
  dpkg-buildpackage -rfakeroot -i.git
to build packages with the change.  This has not been uploaded in any
official packages, I let the package maintainers decide if they want to
support the patch.

b) I had to work around the issue with the media descriptor protocol
sub-field.  In JSCommunicator (using the branch "develop" from JsSIP), I
look at the field in the outgoing and incoming INVITE and change it
to/from the Asterisk format:
https://github.com/opentelecoms-org/jscommunicator/commit/6980f8e1c3311c46154b3840d695f0ddc9c8c8ae

It can now be tested with the links at http://www.sip5060.net/test-calls
and/or from http://www.lumicall.org/drucall - both now appear to work
from Firefox and it appears to maintain compatibility for calls between
JSCommunicator users.

However, I'd like to understand if I really should have the patch/hack
in JSCommunicator at all - should Asterisk be willing to accept SDP
specifying "RTP/SAVPF" alone?  If so, then I can cut out half the
JSCommunicator patch.




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