Hey Flavio! Hope all is well with you (aside from this little SDP issue). Looking forward to seeing you guys at ClueCon 2013. :)<br><br>Are you able to reproduce this issue on latest git HEAD? If so, could you open a ticket at <a href="http://jira.freeswitch.org">jira.freeswitch.org</a> please? That will be the quickest way to get it resolved.<br>
<br>Thanks!<br><br>-Michael<br><br><div class="gmail_quote">On Mon, Feb 25, 2013 at 6:08 AM, Flavio Goncalves <span dir="ltr"><<a href="mailto:flavio@sippulse.com" target="_blank">flavio@sippulse.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">Hi,
<div><br></div><div>I have searched the list, but I couldn't find a final answer for this case. I'm receiving calls from a provider using G729 annex B. </div><div><br></div><div>INVITE</div><div>
<div><font face="arial, sans-serif">a=rtpmap:18 G729/8000.</font></div><div><font face="arial, sans-serif">a=fmtp:18 annexb=yes.</font></div><div><font face="arial, sans-serif">a=rtpmap:0 PCMU/8000.</font></div><div><font face="arial, sans-serif">a=rtpmap:8 PCMA/8000.</font></div>
<div><font face="arial, sans-serif">a=rtpmap:101 telephone-event/8000.</font></div><div><font face="arial, sans-serif">a=fmtp:101 0-16.</font></div><div><font face="arial, sans-serif">a=rtpmap:19 CN/8000.</font></div><div>
<br></div></div><div><span style="font-family:arial,sans-serif;font-size:13.333333969116211px">and FS is answering </span><br></div><div><span style="font-family:arial,sans-serif;font-size:13.333333969116211px"><br>
</span></div><div><span style="font-family:arial,sans-serif;font-size:13.333333969116211px">200OK</span></div><div><div><font face="arial, sans-serif">m=audio 31388 RTP/AVP 18 101 19.</font></div><div><font face="arial, sans-serif">a=rtpmap:18 G729/8000.</font></div>
<div><font face="arial, sans-serif">a=fmtp:18 annexb=yes.</font></div><div><font face="arial, sans-serif">a=rtpmap:101 telephone-event/8000.</font></div><div><font face="arial, sans-serif">a=fmtp:101 0-16.</font></div><div>
<font face="arial, sans-serif">a=rtpmap:19 CN/8000.</font></div><div><font face="arial, sans-serif">a=ptime:20.</font></div><div style="font-family:arial,sans-serif;font-size:13.333333969116211px"><br></div><div style="font-family:arial,sans-serif;font-size:13.333333969116211px">
If freeswitch supports only G729A, shouldn't FS refuse the call with 488 or suppress the line annexb=yes?</div></div><div><span style="font-family:arial,sans-serif;font-size:13.333333969116211px"><br></span></div>
<div><span style="font-family:arial,sans-serif;font-size:13.333333969116211px">The audio quality is really bad when I receive calls in this way. After removing the annexb on the gateway the audio is fine, but unfortunately I can't remove from it from all VoIP providers terminating calls. For outbound calls I simply removed the annexb, but for inbound there is no way to do it. </span></div>
<span class="HOEnZb"><font color="#888888">
<div><span style="font-family:arial,sans-serif;font-size:13.333333969116211px"><br></span></div><div><span style="font-family:arial,sans-serif;font-size:13.333333969116211px">Flavio </span></div><div><span style="font-family:arial,sans-serif;font-size:13.333333969116211px"><br>
</span></div><div><span style="font-family:arial,sans-serif;font-size:13.333333969116211px"><br></span></div></font></span></div>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
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