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

[Dan Pascu]

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)

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