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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