<div dir="ltr">Thanks for this Anthony - I'm still on the stable branch which doesn't seem to support this, but I'll get it upgraded to HEAD and test this & your new patch too.<div><br></div><div>On the topic of codecs, has there been much thought given to video transcoding? I've seen references to a "mod_ffmpeg" floating around, and I've done a bit of transcoding work with libavcodec myself - it seems doable, and the little bit that I've looked at the freeswitch codec examples looks really extensible. If I were to write codecs for H264 and VP8 (and maybe H263+ too), would they be something that you'd try to incorporate into the main project, or would it be better to have as a third-party mod due to the extra hassle of having to build with libav(codec|format|util)?</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Sep 20, 2013 at 9:18 AM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">I'd say its relatively hard. Easier in master which is the only place it would be possible to do but it requires some heavy mods to the media engine and the rtp stack to track all the payloads and switch between them.<div>
<br></div><div>I do think there is another workaround if you set this profile param:</div><div><br></div><div>rtp-autofix-pt=true </div><div><br></div><div><br></div><div>That will detect when its not the one FS thinks and it will reinit the codec.</div>
<div><br></div><div><br></div><div><br></div><div><br></div><div> </div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Sep 19, 2013 at 4:10 PM, Sam Russell <span dir="ltr"><<a href="mailto:sam.h.russell@gmail.com" target="_blank">sam.h.russell@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Thanks for the quick reply Anthony.<div><br></div><div>Do you think this is a hard/easy thing to implement? I'm keen to contribute a bit to freeswitch, and this is functionality that I'd like to write some code for, and I can do what I like with our office PBX so I can test it here too.</div>
</div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Sep 20, 2013 at 8:56 AM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">We currently only support one codec in 200OK The workaround for this is to only offer it PCMU so it has no choice.<div>
<br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Thu, Sep 19, 2013 at 2:52 PM, Sam Russell <span dir="ltr"><<a href="mailto:sam.h.russell@gmail.com" target="_blank">sam.h.russell@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>I've had a couple of calls recently that have been a little weird -<br>
here's the scenario:<br>
<br>
A (local freeswitch) sends invite with PCMU, PCMA as codecs<br>
<br>
B (foreign) sends 200 OK with PCMA, PCMU as codecs<br>
<br>
Freeswitch sees the response and sets audio codec to PCMA, but other<br>
end sends us PCMU (as it was our first choice in this case - sometimes<br>
they choose a random one that both of us support).<br>
<br>
RTP is fine with this - each packet refers to the # of the codec<br>
(either standard or a reference to an rtpmap from the SDP), but<br>
freeswitch doesn't like it - it forwards the PCMA from my soft phone,<br>
but the PCMU coming in from the foreign phone gets dropped by<br>
freeswitch and doesn't get forwarded onto my soft phone (packet trace<br>
shows freeswitch being the black hole).<br>
<br>
Debug output shows freeswitch setting the audio codec, and I suspect<br>
this means it then only accepts RTP with that codec, instead of<br>
handling all codecs that it offered in its invite.<br>
<br>
Is this expected behaviour? I went trawling through RFCs yesterday and<br>
nothing jumped out at me, and I'm seeing this on a couple of call<br>
managers - also sometimes the other end will send H263+ even though<br>
both ends offered H264 as preferred, and freeswitch drops this in the<br>
same manner. I think freeswitch should be handling all codecs that it<br>
offers, especially for video when it only does pass through, so<br>
spinning up instances of extra "codecs" shouldn't create much<br>
overhead.<br>
<br>
Sent from my iPhone<br>
<br></div></div>
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