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

[Nitesh Bansal]

I have another question on the same, even if use the media-proxy or not, i
assume that DTLS message retransmissions need to be handled.
I would like some pointers on how can we handle retransmissions in asterisk.
P.S: I am not an expert on openssl library, so don't know much about how it
functions internally.

Regards,
Nitesh


On Mon, Jan 27, 2014 at 4:17 PM, Nitesh Bansal <nitesh.bansal at gmail.com>wrote:

> Hello everyone,
>
> Contining on the DTLS-SRTP, i need asterisk to be able to retry DTLS
> handshake in case there is no response from the peer for the first
> attempted handshake.
> This is happening in case i use the media-proxy with asterisk, media-proxy
> is sending DTLS data before completing the ICE handshake, so DTLS messages
> are being
> sent to an ICE candidate which is different from final selected ice
> candidate. In this case, i would like asterisk to attempt the DTLS
> handshake after a specific timeout?
> Any pointers on how this can be done ( i can think of scheduling a timer) ?
> P.S: With media-proxy, asterisk sees the de-iced SDP, media-proxy is
> handling the ICE handshake on its own.
>
> Regards,
> Nitesh Bansal
>
>
>
> On Fri, Jan 24, 2014 at 4:22 PM, Daniel Pocock <daniel at pocock.com.au>wrote:
>
>>  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-callsand/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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>
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