<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><br><div><div>On 20 Nov 2014, at 16:54, Kevin Layer <<a href="mailto:layer@franz.com">layer@franz.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div style="font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;"><a href="mailto:ag@ag-projects.com">ag@ag-projects.com</a><span class="Apple-converted-space"> </span>wrote:<br><br><blockquote type="cite"><blockquote type="cite">As Dan explained, the problem is not in Blink but in the remote<br>server. The server should not send a video stream at all or if it<br>sends one it must be syntactically correct.<span class="Apple-converted-space"> </span><br><br>I am not sure what we can fix in this respect but you should ask you<br>provider to make a fix their server as their INVITE is broken.<br></blockquote></blockquote><br>Counterpoint: Just because the video stream could not be negotiated<br>does not mean that it must reject the invite entirely.  </div></blockquote><div><br></div><div>This is true. It would be best to discard the failed stream and accept the audio only. </div><div><br></div><div><br></div><blockquote type="cite"><div style="font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">How do they<br>explain the successful operation of all of our other devices which<br>work properly with this version of Asterisk?  Are they all out of<br>spec?<br></div></blockquote><div><br></div><div>For this we have the theory of two broken devices that work well together until one that works right comes along ;-)</div><br><blockquote type="cite"><div style="font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">Further, rfc3984 (and the newer rfc6184) indicates that<br>profile-level-id is an optional parameter.<br><br>And, it's not like we have some crazy, off-brand server.  We're using<br>the LTS branch of Asterisk without modification.<br></div></blockquote><div><br></div><div>This is really crazy.</div><br><blockquote type="cite"><div style="font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">Kevin<br><br><blockquote type="cite"><blockquote type="cite">Adrian<br><br>On 20 Nov 2014, at 14:07, Kevin Layer <<a href="mailto:layer@franz.com">layer@franz.com</a>> wrote:<br><br>   Dan Pascu wrote:<br><br>               The INVITE you receive from asterisk has an<br>           incomplete/invalid video stream specification. A line<br>           similar to this one (that you can see in the 200 OK that<br>           Blink sends), must also be present in the INVITE from<br>           asterisk:<br><br>           a=fmtp:99 profile-level-id=42001f;packetization-mode=0<br><br>           without it, the codec cannot be negotiated as it has no<br>           idea what parameters will be used and as a result you see<br>           the error:<br><br>           Could not initialize RTP for video session: Codec internal<br>           creation<br>           error (PJMEDIA_CODEC_EFAILED) (500)<br><br>   I see this in the error I reported yesterday.<br><br>   I stupidly upgraded my home machine to 4.1.0 before trying the old<br>   version. Now incoming calls are broken for me at home, too!<br><br>   Adrian, do you know what the issue is? Is a fix coming soon? I<br>   have<br>   to switch to another app because I can't receive calls. Thanks.<br><br><br>           On 19 Nov 2014, at 23:08, Kevin Layer wrote:<br><br>                       Here's the SIP log, for another call that<br>               behaved the same way:<br><br>               RECEIVED: Packet 70, +0:12:51.767797<br>               2014-11-19 13:05:10.051697: xxx.xxx.xxx.5:5060 -(SIP<br>               over UDP)-><br>               xxx.xxx.xxx.149:57388<br>               INVITE <a href="sip:26951487@xxx.xxx.xxx.149:57388">sip:26951487@xxx.xxx.xxx.149:57388</a> SIP/2.0<br>               Via: SIP/2.0/UDP<br>               xxx.xxx.xxx.5:5060;branch=z9hG4bK3abbdff1;rport<br>               Max-Forwards: 70<br>               From: "Mr. Foo Bar"<br>               <<a href="sip:127@xxx.xxx.xxx.5">sip:127@xxx.xxx.xxx.5</a>>;tag=as23444ac2<br>               To: <<a href="sip:26951487@xxx.xxx.xxx.149:57388">sip:26951487@xxx.xxx.xxx.149:57388</a>><br>               Contact: <<a href="sip:127@xxx.xxx.xxx.5:5060">sip:127@xxx.xxx.xxx.5:5060</a>><br>               Call-ID:<br>               <a href="mailto:25f99afe01f1503543905cc47c493627@xxx.xxx.xxx">25f99afe01f1503543905cc47c493627@xxx.xxx.xxx</a>.5:5060<br>               CSeq: 102 INVITE<br>               User-Agent: FPBX-2.10.0(1.8.11)<br>               Date: Wed, 19 Nov 2014 21:05:10 GMT<br>               Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,<br>               SUBSCRIBE, NOTIFY,<br>               INFO, PUBLISH<br>               Supported: replaces, timer<br>               Content-Type: application/sdp<br>               Content-Length: 417<br><br>               v=0<br>               o=root 291324575 291324575 IN IP4 xxx.xxx.xxx.5<br>               s=Asterisk PBX 1.8.11-cert7<br>               c=IN IP4 xxx.xxx.xxx.5<br>               b=CT:384<br>               t=0 0<br>               m=audio 15408 RTP/AVP 0 8 3 101<br>               a=rtpmap:0 PCMU/8000<br>               a=rtpmap:8 PCMA/8000<br>               a=rtpmap:3 GSM/8000<br>               a=rtpmap:101 telephone-event/8000<br>               a=fmtp:101 0-16<br>               a=ptime:20<br>               a=sendrecv<br>               m=video 12210 RTP/AVP 34 98 99<br>               a=rtpmap:34 H263/90000<br>               a=rtpmap:98 h263-1998/90000<br>               a=rtpmap:99 H264/90000<br>               a=sendrecv<br><br>               SENDING: Packet 73, +0:12:52.992536<br>               2014-11-19 13:05:11.276436: xxx.xxx.xxx.149:57388 -<br>               (SIP over UDP)-><br>               xxx.xxx.xxx.5:5060<br>               SIP/2.0 200 OK<br>               Via: SIP/2.0/UDP<br>               xxx.xxx.xxx.5:5060;rport=5060;received=xxx.xxx.xxx.5;branch=z9hG4bK3abbdff1<br>               Call-ID:<br>               <a href="mailto:25f99afe01f1503543905cc47c493627@xxx.xxx.xxx">25f99afe01f1503543905cc47c493627@xxx.xxx.xxx</a>.5:5060<br>               From: "Mr. Foo Bar"<br>               <<a href="sip:127@xxx.xxx.xxx.5">sip:127@xxx.xxx.xxx.5</a>>;tag=as23444ac2<br>               To:<br>               <<a href="sip:26951487@xxx.xxx.xxx.149">sip:26951487@xxx.xxx.xxx.149</a>>;tag=KHPrL2-2MnSuUCRk10Vf6KJ7e4cBe0Gl<br>               CSeq: 102 INVITE<br>               Server: Blink Pro 4.1.0 (MacOSX)<br>               Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE,<br>               CANCEL, UPDATE,<br>               MESSAGE, REFER<br>               Contact: <<a href="sip:26951487@xxx.xxx.xxx.149:57388">sip:26951487@xxx.xxx.xxx.149:57388</a>><br>               Supported: 100rel, replaces, norefersub, gruu<br>               Content-Type: application/sdp<br>               Content-Length: 582<br><br>               v=0<br>               o=- 3625419911 3625419912 IN IP4 xxx.xxx.xxx.149<br>               s=Blink Pro 4.1.0 (MacOSX)<br>               t=0 0<br>               m=audio 50012 RTP/AVP 0 101<br>               c=IN IP4 xxx.xxx.xxx.149<br>               a=rtcp:50013<br>               a=rtpmap:0 PCMU/8000<br>               a=rtpmap:101 telephone-event/8000<br>               a=fmtp:101 0-16<br>               a=zrtp-hash:1.10<br>               2d96ee26bf5ef5ee6b0a13968f12dedfb64f7954c8d4e6e7b625af332ec32613<br>               a=sendrecv<br>               m=video 50014 RTP/AVP 99<br>               c=IN IP4 xxx.xxx.xxx.149<br>               b=TIAS:4000000<br>               a=rtcp:50015<br>               a=zrtp-hash:1.10<br>               6def416cd947a6cc6f69b0acaf62816f97f35224943b81dd8c615a0d4a87ddcc<br>               a=sendrecv<br>               a=rtpmap:99 H264/90000<br>               a=fmtp:99 profile-level-id=42001f;packetization-mode=0<br><br><br>           --<br>           Dan<br><br><br><br><br>           _______________________________________________<br>           Blink mailing list<br>           <a href="mailto:Blink@lists.ag-projects.com">Blink@lists.ag-projects.com</a><br>           <a href="http://lists.ag-projects.com/mailman/listinfo/blink">http://lists.ag-projects.com/mailman/listinfo/blink</a><br><br>   _______________________________________________<br>   Blink mailing list<br>   <a href="mailto:Blink@lists.ag-projects.com">Blink@lists.ag-projects.com</a><br>   <a href="http://lists.ag-projects.com/mailman/listinfo/blink">http://lists.ag-projects.com/mailman/listinfo/blink</a><br><br>--<br>Adrian<br><br><br>_______________________________________________<br>Blink mailing list<br><a href="mailto:Blink@lists.ag-projects.com">Blink@lists.ag-projects.com</a><br><a href="http://lists.ag-projects.com/mailman/listinfo/blink">http://lists.ag-projects.com/mailman/listinfo/blink</a><br></blockquote></blockquote>_______________________________________________<br>Blink mailing list<br><a href="mailto:Blink@lists.ag-projects.com">Blink@lists.ag-projects.com</a><br><a href="http://lists.ag-projects.com/mailman/listinfo/blink">http://lists.ag-projects.com/mailman/listinfo/blink</a></div></blockquote></div><br><div>
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