[Blink] New Blink Pro for OSX version 4.1.0 with ZRTP end-to-end audio/video encryption

ag at ag-projects.com ag at ag-projects.com
Thu Nov 20 20:55:53 CET 2014


On 20 Nov 2014, at 16:54, Kevin Layer <layer at franz.com> wrote:

> ag at ag-projects.com wrote:
> 
>>> As Dan explained, the problem is not in Blink but in the remote
>>> server. The server should not send a video stream at all or if it
>>> sends one it must be syntactically correct. 
>>> 
>>> I am not sure what we can fix in this respect but you should ask you
>>> provider to make a fix their server as their INVITE is broken.
> 
> Counterpoint: Just because the video stream could not be negotiated
> does not mean that it must reject the invite entirely.  

This is true. It would be best to discard the failed stream and accept the audio only. 


> How do they
> explain the successful operation of all of our other devices which
> work properly with this version of Asterisk?  Are they all out of
> spec?

For this we have the theory of two broken devices that work well together until one that works right comes along ;-)

> Further, rfc3984 (and the newer rfc6184) indicates that
> profile-level-id is an optional parameter.
> 
> And, it's not like we have some crazy, off-brand server.  We're using
> the LTS branch of Asterisk without modification.

This is really crazy.

> Kevin
> 
>>> Adrian
>>> 
>>> On 20 Nov 2014, at 14:07, Kevin Layer <layer at franz.com> wrote:
>>> 
>>>    Dan Pascu wrote:
>>> 
>>>                The INVITE you receive from asterisk has an
>>>            incomplete/invalid video stream specification. A line
>>>            similar to this one (that you can see in the 200 OK that
>>>            Blink sends), must also be present in the INVITE from
>>>            asterisk:
>>> 
>>>            a=fmtp:99 profile-level-id=42001f;packetization-mode=0
>>> 
>>>            without it, the codec cannot be negotiated as it has no
>>>            idea what parameters will be used and as a result you see
>>>            the error:
>>> 
>>>            Could not initialize RTP for video session: Codec internal
>>>            creation
>>>            error (PJMEDIA_CODEC_EFAILED) (500)
>>> 
>>>    I see this in the error I reported yesterday.
>>> 
>>>    I stupidly upgraded my home machine to 4.1.0 before trying the old
>>>    version. Now incoming calls are broken for me at home, too!
>>> 
>>>    Adrian, do you know what the issue is? Is a fix coming soon? I
>>>    have
>>>    to switch to another app because I can't receive calls. Thanks.
>>> 
>>> 
>>>            On 19 Nov 2014, at 23:08, Kevin Layer wrote:
>>> 
>>>                        Here's the SIP log, for another call that
>>>                behaved the same way:
>>> 
>>>                RECEIVED: Packet 70, +0:12:51.767797
>>>                2014-11-19 13:05:10.051697: xxx.xxx.xxx.5:5060 -(SIP
>>>                over UDP)->
>>>                xxx.xxx.xxx.149:57388
>>>                INVITE sip:26951487 at xxx.xxx.xxx.149:57388 SIP/2.0
>>>                Via: SIP/2.0/UDP
>>>                xxx.xxx.xxx.5:5060;branch=z9hG4bK3abbdff1;rport
>>>                Max-Forwards: 70
>>>                From: "Mr. Foo Bar"
>>>                <sip:127 at xxx.xxx.xxx.5>;tag=as23444ac2
>>>                To: <sip:26951487 at xxx.xxx.xxx.149:57388>
>>>                Contact: <sip:127 at xxx.xxx.xxx.5:5060>
>>>                Call-ID:
>>>                25f99afe01f1503543905cc47c493627 at xxx.xxx.xxx.5:5060
>>>                CSeq: 102 INVITE
>>>                User-Agent: FPBX-2.10.0(1.8.11)
>>>                Date: Wed, 19 Nov 2014 21:05:10 GMT
>>>                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>>>                SUBSCRIBE, NOTIFY,
>>>                INFO, PUBLISH
>>>                Supported: replaces, timer
>>>                Content-Type: application/sdp
>>>                Content-Length: 417
>>> 
>>>                v=0
>>>                o=root 291324575 291324575 IN IP4 xxx.xxx.xxx.5
>>>                s=Asterisk PBX 1.8.11-cert7
>>>                c=IN IP4 xxx.xxx.xxx.5
>>>                b=CT:384
>>>                t=0 0
>>>                m=audio 15408 RTP/AVP 0 8 3 101
>>>                a=rtpmap:0 PCMU/8000
>>>                a=rtpmap:8 PCMA/8000
>>>                a=rtpmap:3 GSM/8000
>>>                a=rtpmap:101 telephone-event/8000
>>>                a=fmtp:101 0-16
>>>                a=ptime:20
>>>                a=sendrecv
>>>                m=video 12210 RTP/AVP 34 98 99
>>>                a=rtpmap:34 H263/90000
>>>                a=rtpmap:98 h263-1998/90000
>>>                a=rtpmap:99 H264/90000
>>>                a=sendrecv
>>> 
>>>                SENDING: Packet 73, +0:12:52.992536
>>>                2014-11-19 13:05:11.276436: xxx.xxx.xxx.149:57388 -
>>>                (SIP over UDP)->
>>>                xxx.xxx.xxx.5:5060
>>>                SIP/2.0 200 OK
>>>                Via: SIP/2.0/UDP
>>>                xxx.xxx.xxx.5:5060;rport=5060;received=xxx.xxx.xxx.5;branch=z9hG4bK3abbdff1
>>>                Call-ID:
>>>                25f99afe01f1503543905cc47c493627 at xxx.xxx.xxx.5:5060
>>>                From: "Mr. Foo Bar"
>>>                <sip:127 at xxx.xxx.xxx.5>;tag=as23444ac2
>>>                To:
>>>                <sip:26951487 at xxx.xxx.xxx.149>;tag=KHPrL2-2MnSuUCRk10Vf6KJ7e4cBe0Gl
>>>                CSeq: 102 INVITE
>>>                Server: Blink Pro 4.1.0 (MacOSX)
>>>                Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE,
>>>                CANCEL, UPDATE,
>>>                MESSAGE, REFER
>>>                Contact: <sip:26951487 at xxx.xxx.xxx.149:57388>
>>>                Supported: 100rel, replaces, norefersub, gruu
>>>                Content-Type: application/sdp
>>>                Content-Length: 582
>>> 
>>>                v=0
>>>                o=- 3625419911 3625419912 IN IP4 xxx.xxx.xxx.149
>>>                s=Blink Pro 4.1.0 (MacOSX)
>>>                t=0 0
>>>                m=audio 50012 RTP/AVP 0 101
>>>                c=IN IP4 xxx.xxx.xxx.149
>>>                a=rtcp:50013
>>>                a=rtpmap:0 PCMU/8000
>>>                a=rtpmap:101 telephone-event/8000
>>>                a=fmtp:101 0-16
>>>                a=zrtp-hash:1.10
>>>                2d96ee26bf5ef5ee6b0a13968f12dedfb64f7954c8d4e6e7b625af332ec32613
>>>                a=sendrecv
>>>                m=video 50014 RTP/AVP 99
>>>                c=IN IP4 xxx.xxx.xxx.149
>>>                b=TIAS:4000000
>>>                a=rtcp:50015
>>>                a=zrtp-hash:1.10
>>>                6def416cd947a6cc6f69b0acaf62816f97f35224943b81dd8c615a0d4a87ddcc
>>>                a=sendrecv
>>>                a=rtpmap:99 H264/90000
>>>                a=fmtp:99 profile-level-id=42001f;packetization-mode=0
>>> 
>>> 
>>>            --
>>>            Dan
>>> 
>>> 
>>> 
>>> 
>>>            _______________________________________________
>>>            Blink mailing list
>>>            Blink at lists.ag-projects.com
>>>            http://lists.ag-projects.com/mailman/listinfo/blink
>>> 
>>>    _______________________________________________
>>>    Blink mailing list
>>>    Blink at lists.ag-projects.com
>>>    http://lists.ag-projects.com/mailman/listinfo/blink
>>> 
>>> --
>>> Adrian
>>> 
>>> 
>>> _______________________________________________
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>>> Blink at lists.ag-projects.com
>>> http://lists.ag-projects.com/mailman/listinfo/blink
> _______________________________________________
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--
Adrian



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