[Blink] New Blink Pro for OSX version 4.1.0 with ZRTP end-to-end audio/video encryption
Kevin Layer
layer at franz.com
Thu Nov 20 22:09:32 CET 2014
ag at ag-projects.com wrote:
>> 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.
I can't tell if you're being flip. Can a fix please made? Because
the two copies of Blink Pro I bought are completely useless to me now.
Thanks.
Kevin
>>
>> 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
>>
>>
>>
>>
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>> --
>> Adrian
>>
>>
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>> --
>> Adrian
>>
>>
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