<div dir="ltr"><div class="gmail_extra"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"></blockquote>Hi Joshua,<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>Thank you for looking into this.<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>Their response IS based on traces I've sent them. Attached is such trace in text format (server IP has been changed to 111.111.111.111). Some repeating RTP packets has been truncated. </div><div class="gmail_quote">You can see that after the 200 OK SSRC is sent from the server to the phone as '0x0'. The same has happened with G729 codec.<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>Let me know if you need the full trace or anything else from my side.<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>I should also mention that this is Asterisk version 1.8.12.1 <br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>Thank you<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>Harel<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>------------------------------<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
</blockquote><br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>On Tue, Nov 14, 2017, at 09:14 AM, Harel Cohen wrote:<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> Hello,<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> I have a problem where on an outgoing call a Grandstream phone<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> (GXP2130) closes the incoming voice stream about 1 second into the<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> call (the remote party hears the Grandstream, the Grandstream doesn't<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> hear thr remote party). I have verified with logs and traces that this<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> is not a NAT issue or any other network-related problem. All incoming<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> RTP packets arrive at the phone on the correct port etc. as declared in the SDP.<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> I opened a ticket with Grandstream and they replied: "<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>><br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> *the phone starts receiving RTP with SSRC =0x0 which is wrong".*<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
</blockquote>><br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> Is this an Asterisk problem or the phones? Is this something that can<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>> be fixed on the Asterisk side?<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote><br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>Asterisk would be sending the RTP to the Grandstream. I'd suggest getting a packet capture using tcpdump or wireshark to confirm what they've said though. I just looked at the code and I don't see a way that we'd ever have the SSRC be 0.<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
</blockquote><br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>Cheers,<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
</blockquote><br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>--<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>Joshua Colp<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>Digium, Inc. | Senior Software Developer<br><blockquote class="gmail_quote" style="margin:0px 0.8ex;border-left:1px solid rgb(204,204,204);border-right:1px solid rgb(204,204,204);padding-left:1ex;padding-right:1ex"></blockquote>445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: <a href="http://www.digium.com" rel="noreferrer" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" rel="noreferrer" target="_blank">www.asterisk.org</a><br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
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