<div dir="ltr"><div class="gmail_default" style="font-family:arial narrow,sans-serif"><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long <span dir="ltr"><<a href="mailto:kevin.long@haloprivacy.com" target="_blank">kevin.long@haloprivacy.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="auto"><div>Hi George the patch was from here , you wrote it I believe . I pulled asterisk 13 from git, apply this patch which fixed RTP issue , but I think tla transport issue came back for me . </div><div><br></div><div><a href="https://gerrit.asterisk.org/#/c/2346/" target="_blank">https://gerrit.asterisk.org/#/c/2346/</a></div></div></blockquote><div><br></div><div><div class="gmail_default" style="font-family:'arial narrow',sans-serif">Oh, that one, OK. It should be merged now so if you 'git pull' on 13 now, you should get it. The transport re-use issue was in pjproject so is it possible that you're not compiling against the latest trunk?</div><br></div><div><br></div><div><br></div><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto"><div><br></div><div>Thank you<br><br>Sent from my iPhone</div><div><div class="h5"><div><br>On Mar 4, 2016, at 12:01 AM, George Joseph <<a href="mailto:george.joseph@fairview5.com" target="_blank">george.joseph@fairview5.com</a>> wrote:<br><br></div><blockquote type="cite"><div><div dir="ltr"><div style="font-family:'arial narrow',sans-serif"><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <span dir="ltr"><<a href="mailto:kevin.long@haloprivacy.com" target="_blank">kevin.long@haloprivacy.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><br>
Thanks George I appreciate the info . Being able to see what codec is in use for call in progress is very handy sometimes.<br>
<br>
As far as the RTP stats goes, I see there is some info with “rtp” and “rtcp” commands which can be useful for troubleshooting. A running tally of # packets or bandwidth used would be awesome in along with the codec in "pjsip show channels" or something like that.<br>
<br>
<br>
Im not certain, but I think the TLS signalling problem from this email may be happening to me again after patching for another pjsip/NAT issue which was with the external_media_address not working and the internal IP being sent in the SDP from asterisk - I applied this patch to the codebase and recompiled I am seeing the TLS “new transport” issue again , I think.<br></blockquote><div><br></div><div><div style="font-family:'arial narrow',sans-serif">I've lost track of who's applying what patches to which codebase. :)</div><div style="font-family:'arial narrow',sans-serif"><br></div><div style="font-family:'arial narrow',sans-serif">Which patch did you apply for "<span style="font-family:arial,sans-serif">external_media_address not working"?</span></div></div><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">
<br>
Regards,<br>
<br>
Kevin Long<br>--<br>
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