[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
>>>
>>>
>>>
>>>
>>> _______________________________________________
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>>>
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>>> Blink at lists.ag-projects.com
>>> http://lists.ag-projects.com/mailman/listinfo/blink
>>>
>>> --
>>> Adrian
>>>
>>>
>>> _______________________________________________
>>> Blink mailing list
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>>> http://lists.ag-projects.com/mailman/listinfo/blink
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--
Adrian
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