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<div class="moz-cite-prefix">On 24/01/14 10:59, Lorenzo Miniero
wrote:<br>
</div>
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cite="mid:CAHEg3BLyFyAeamShevpi+i3kz9dP2w428w0YXz8gnbsVd39uGQ@mail.gmail.com"
type="cite">
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<div dir="ltr">Hi Daniel,
<div><br>
</div>
<div>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:</div>
<div><br>
</div>
<div><a moz-do-not-send="true"
href="https://issues.asterisk.org/jira/browse/ASTERISK-22961">https://issues.asterisk.org/jira/browse/ASTERISK-22961</a><br>
</div>
<div><br>
</div>
<div>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.</div>
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</blockquote>
<br>
Thanks for this, I've tested with it<br>
<br>
Two things were necessary for success with Firefox:<br>
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<br>
<a class="moz-txt-link-freetext" href="http://anonscm.debian.org/gitweb/?p=pkg-voip/asterisk.git;a=shortlog;h=refs/heads/dtls-srtp-patch">http://anonscm.debian.org/gitweb/?p=pkg-voip/asterisk.git;a=shortlog;h=refs/heads/dtls-srtp-patch</a><br>
Anybody wanting to test can clone from there and then<br>
dpkg-buildpackage -rfakeroot -i.git <br>
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.<br>
<br>
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:<br>
<a class="moz-txt-link-freetext" href="https://github.com/opentelecoms-org/jscommunicator/commit/6980f8e1c3311c46154b3840d695f0ddc9c8c8ae">https://github.com/opentelecoms-org/jscommunicator/commit/6980f8e1c3311c46154b3840d695f0ddc9c8c8ae</a><br>
<br>
It can now be tested with the links at
<a class="moz-txt-link-freetext" href="http://www.sip5060.net/test-calls">http://www.sip5060.net/test-calls</a> and/or from
<a class="moz-txt-link-freetext" href="http://www.lumicall.org/drucall">http://www.lumicall.org/drucall</a> - both now appear to work from
Firefox and it appears to maintain compatibility for calls between
JSCommunicator users.<br>
<br>
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.<br>
<br>
<br>
<br>
<br>
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