I believe that is exactly the way to use it.<div>-MC<br><br><div class="gmail_quote">On Tue, Sep 13, 2011 at 4:53 PM, Avi Marcus <span dir="ltr"><<a href="mailto:avi@avimarcus.net">avi@avimarcus.net</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"><div>OK then!</div>Looking at my call logs, it seems to mostly happen with one particular carrier. Can I set <span style="font-family:monospace;font-size:13px;line-height:14px;white-space:pre-wrap">verbose_sdp=true </span><span style="font-family:Verdana, Arial, Helvetica, sans-serif">in the bridge string? e.g. "[verbose_sdp=true]sofia/gateway/... "?</span><div>
<font face="Verdana, Arial, Helvetica, sans-serif"><br></font></div><div><font face="Verdana, Arial, Helvetica, sans-serif">Thanks,<br></font><font color="#888888"><div dir="ltr"><span style="font-family:Verdana, Arial, Helvetica, sans-serif"><span style="font-size:small">-Avi</span></span></div>
</font><div><div></div><div class="h5">
<br><br><div class="gmail_quote">On Tue, Sep 13, 2011 at 6:28 PM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
For the record, line #107 contains the codecs (and other stuff) that FS offered:<div><br></div><div><span style="color:rgb(51, 51, 51);font-family:'Courier New', Courier, monospace;font-size:11px;background-color:rgb(255, 255, 255)">m=audio <span>16480</span> RTP/AVP <span>8</span> <span>0</span> <span>9</span> <span>101</span> <span>13</span></span><br>
<br>The "8 0 9 101 13" represent the codecs (and other stuff). What do those numbers mean? This wiki page has the list:</div><div><br></div><div><a href="http://wiki.freeswitch.org/wiki/RTP_payload_list" target="_blank">http://wiki.freeswitch.org/wiki/RTP_payload_list</a></div>
<div><br></div><div>The SDP spec says that for non-dynamic payload numbers you are NOT required to have the "a=rtpmap" lines. This makes sense because those payload numbers don't change meaning until you get up to 96 and higher. The issue is that some devices are broken and require the "a=rtpmap" lines for the payload types below number 96. The verbose_sdp chan var mentioned by KK tells FS to add all those a=rtpmap lines for the non-dynamic payload types. You should only set this var when absolutely necessary because those verbose SDP's can cause the packets to be larger than the allowable MTU size.</div>
<div><br></div><div>Dontcha just *love* SIP/SDP interop? :)</div><div><br></div><div><font color="#888888">-MC <br><br></font><div class="gmail_quote"><div><div></div><div>On Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <span dir="ltr"><<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</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></div><div><div dir="ltr"><div>I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't show PCMU in it - did FS not offer any codecs?</div>
<div>
FS log: <a href="http://pastebin.freeswitch.org/17316" target="_blank">http://pastebin.freeswitch.org/17316</a></div><div>PCAP: <a href="http://ge.tt/95tFji7" target="_blank">http://ge.tt/95tFji7</a></div>
<div><br></div><div>Can someone tell me what's going on here?</div><div><br></div><font color="#888888"><div><span style="font-family:Verdana, Arial, Helvetica, sans-serif">-Avi</span></div>
</font></div>
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