<div dir="ltr">Hi Olle,<div><br></div><div>answering inline.<br><div class="gmail_extra"><br><br><div class="gmail_quote">2013/5/27 Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>></span><br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div style="word-wrap:break-word">Thank you, Lorenzo!<div>
<br><div>A few comments:</div><div><br></div><div>We need to find a better way to handle video update requests - your code change in chan-sip is not very<div>modular. We should update the existing <span style="font-family:Consolas,'Liberation Mono',Courier,monospace;font-size:12px;line-height:18px;white-space:pre-wrap;background-color:rgb(255,221,221)">ast_rtcp_send_h261fur() function if that one doesn't work.</span></div>
<div><font face="Consolas, Liberation Mono, Courier, monospace"><span style="font-size:12px;line-height:18px;white-space:pre-wrap"><br></span></font></div></div></div></div></blockquote><div><br></div><div style>Actually ast_rtcp_send_h261fur() does not exist anywhere in the code: it was already commented in chan_sip. What I added was simply a "hack" to send a RTCP FIR message in the VP8 case, rather than a SIP INFO that was the default behaviour (the transmit_info_with_vidupdate that is there). I did it this way to minimize the impact on the existing code, but I'll welcome any indication on better ways to procede!</div>
<div style> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div style="word-wrap:break-word"><div><div>
<div><font face="Consolas, Liberation Mono, Courier, monospace"><span style="font-size:12px;line-height:18px;white-space:pre-wrap"></span></font></div><div><font face="Consolas, Liberation Mono, Courier, monospace"><span style="font-size:12px;line-height:18px;white-space:pre-wrap">Also, your adding this:</span></font></div>
<div><span style="font-family:Consolas,'Liberation Mono',Courier,monospace;font-size:12px;line-height:18px;white-space:pre-wrap;background-color:rgb(221,255,221)">ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d/2\r\n", rtp_code, mime, rate);</span></div>
<div><span style="font-family:Consolas,'Liberation Mono',Courier,monospace;font-size:12px;line-height:18px;white-space:pre-wrap;background-color:rgb(221,255,221)"><br></span></div><div><span style="font-family:Consolas,'Liberation Mono',Courier,monospace;font-size:12px;line-height:18px;white-space:pre-wrap;background-color:rgb(221,255,221)">I don't think Asterisk supports parsing that on incoming SDP. Have you tested this?</span></div>
<div><br></div></div></div></div></blockquote><div><br></div><div style>The append you see is mandated by the current specification:</div><div style><br></div><div style> <a href="http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-00#section-6.2">http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-00#section-6.2</a></div>
<div style><div> The RTP clock rate in "a=rtpmap" MUST be 48000 and the number of channels MUST be 2.</div></div><div style><br></div><div style>Asterisk seems to parse this fine, though (both browsers do this, I can't remember if PhonerLite does as well), as Opus is correctly negotiated. I can't remember any issue when checking with "sip set debug on" either. My guess is that in the "rtpmap: %30u %127[^/]/%30u" sscanf it is simply ignored.<br>
</div><div style><br></div><div style> <br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div style="word-wrap:break-word">
<div><div><div></div><div><span style="font-family:Consolas,'Liberation Mono',Courier,monospace;font-size:12px;line-height:18px;white-space:pre-wrap;background-color:rgb(221,255,221)">Apologize for the colors, it's from cut/paste.</span></div>
<span class=""><font color="#888888"><div><span style="font-family:Consolas,'Liberation Mono',Courier,monospace;font-size:12px;line-height:18px;white-space:pre-wrap;background-color:rgb(221,255,221)"><br></span></div>
</font></span></div></div></div></blockquote><div><br></div><div style>Long live colors! :-)</div><div style>Lorenzo</div><div style> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div style="word-wrap:break-word"><div><div><span class=""><font color="#888888"><div><span style="font-family:Consolas,'Liberation Mono',Courier,monospace;font-size:12px;line-height:18px;white-space:pre-wrap;background-color:rgb(221,255,221)"></span></div>
<div><font face="Consolas, Liberation Mono, Courier, monospace"><span style="font-size:12px;line-height:18px;white-space:pre-wrap">/O</span></font></div></font></span><div><font face="Consolas, Liberation Mono, Courier, monospace"><span style="font-size:12px;line-height:18px;white-space:pre-wrap"><br>
</span></font><div><div class="im"><div>27 maj 2013 kl. 16:09 skrev Lorenzo Miniero <<a href="mailto:lminiero@gmail.com" target="_blank">lminiero@gmail.com</a>>:</div><br></div><div><div class="h5"><blockquote type="cite">
<div dir="ltr">Dear all,<div><br></div><div>I've just published the patch on github:</div><div><br></div><div><a href="https://github.com/meetecho/asterisk-opus" target="_blank">https://github.com/meetecho/asterisk-opus</a><br>
</div><div><br></div><div>The README should be quite self explainatory, but if you need any additional info feel free to ask me.</div><div>Any feedback will be more than welcome!</div><div><br></div>
<div>Lorenzo</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/5/26 Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>></span><br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<br>
26 maj 2013 kl. 12:08 skrev Hans Witvliet <<a href="mailto:asterisk@a-domani.nl" target="_blank">asterisk@a-domani.nl</a>>:<br>
<div><br>
> Seems my mesage didn't reach the list...<br>
> (could me my end of the list that's failing)<br>
</div>This is not about software logic, it's about legal issues and we can't take that discussion here, it's for<br>
Digium to make by themselves and we just have to respect their decision, whatever we think<br>
about it.<br>
<br>
We can create a community distribution based on GPL and we in the community can<br>
encourage as many as possible to use it - but it's their decision about the type of legal<br>
risc they consider it to be.<br>
<br>
Hopefully the issues that stop Digium from supporting it today will diminish so that<br>
Opus and VP8 can be included in the "official" distribution at some point - and then<br>
everything should be ready for some fast action.<br>
<br>
Digium has done everything they can for Asterisk - and more.<br>
This time it's the community that has to prove that we can handle the situation ;-)<br>
<span><font color="#888888"><br>
/O<br>
</font></span><div><div><br>
> Hans<br>
><br>
> -----Original Message-----<br>
> From: Hans Witvliet <<a href="mailto:asterisk@a-domani.nl" target="_blank">asterisk@a-domani.nl</a>><br>
> To: <a href="mailto:asterisk-dev@lists.digium.com" target="_blank">asterisk-dev@lists.digium.com</a><br>
> Subject: Re: [asterisk-dev] Opus and VP8<br>
> Date: Sat, 25 May 2013 12:19:13 +0200<br>
><br>
> -----Original Message-----<br>
> From: Olle E. Johansson <<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>><br>
> Reply-to: Asterisk Developers Mailing List<br>
> <<a href="mailto:asterisk-dev@lists.digium.com" target="_blank">asterisk-dev@lists.digium.com</a>><br>
> To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com" target="_blank">asterisk-dev@lists.digium.com</a>><br>
> Cc: Olle E. Johansson <<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>><br>
> Subject: Re: [asterisk-dev] Opus and VP8<br>
> Date: Fri, 24 May 2013 13:26:29 +0200<br>
><br>
><br>
> 24 maj 2013 kl. 12:51 skrev Lorenzo Miniero <<a href="mailto:lminiero@gmail.com" target="_blank">lminiero@gmail.com</a>>:<br>
><br>
>> PS: a few months ago I also talked, on the #asterisk-dev IRC, about<br>
>> the support I added for both Opus (transcoding) and VP8 (passthrough)<br>
>> in Asterisk, codecs that are currently the default ones used in<br>
>> WebRTC. I checked whether there was an interest in a patch for them,<br>
>> but at the time there were some concerns about the copyright status of<br>
>> Opus that prevented it to be considered for integration in Asterisk.<br>
>> Has this situation changed in the meanwhile? I can open a separate<br>
>> thread for this if needed.<br>
>><br>
> Lorenzo,<br>
><br>
><br>
> Good seeing you here!<br>
><br>
><br>
> Due to legal issues I don't think Digium can accept a contribution of<br>
> Opus and VP8 in the svn repositories today.<br>
><br>
><br>
> I would encourage you, if you have these patches, to publish them on a<br>
> web site like github or sourceforge so w all can help you test it. I<br>
> really would like for these to be available for the community in an easy<br>
> form.<br>
><br>
><br>
> Some things can be done in Asterisk though and that's the code points<br>
> for pass through media. I don't think that would cause any legal<br>
> issues.<br>
><br>
><br>
> Hi Olle,<br>
><br>
> I understand that companies like Digium are very carefully with regards<br>
> to legal aspects, but how come that another USA-based company can<br>
> use/ship vp8 freely (linphone). The European based company that<br>
> builds/distribute Jitsi also ships it in their latest version:<br>
><br>
> Linphone:<br>
> Audio with the following codecs: speex (narrow band and wideband), G711<br>
> (ulaw,alaw), GSM, G722. Through additionals plugins, it also supports<br>
> AMR-NB, SILK, G729 and iLBC.<br>
> Video with codecs: VP8 (WebM), H263, H263-1998, MPEG4, theora and H264<br>
> (thanks to a plugin based on x264), with resolutions from QCIF(176x144)<br>
> to SVGA(800x600) provided that network<br>
><br>
> Jitsi:<br>
> "Among the most prominent new features you will find quality multi-party<br>
> video conferences for XMPP, audio device hot-plugging, support for<br>
> Outlook presence and calls, an overhauled user interface and support for<br>
> the Opus and VP8 audio/video codec. You can download the new version at<br>
> the following location: <a href="https://download.jitsi.org/" target="_blank">https://download.jitsi.org/</a>"<br>
><br>
><br>
><br>
><br>
<br>
</div></div></blockquote></div><br></div>
</blockquote></div></div></div><br></div></div></div></div></blockquote></div><br></div></div></div>