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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Vivek,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The SIP port is set to the standard port 5060. The RTP ports as
far as I know are random ephemeral ports between 63000 and 64000.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I can change the port range on the media server, asterisk and the
device, but neither seems to help.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>My diagram below is probably misleading. The RTP traffic flow
that I see is as follows (one way traffic into Asterisk)<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>SIP Phone <---> Media Gateway <b>---></b> Asterisk <b><---</b>
SIP Phone<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Ryan<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Vivek
Shrivastava<br>
<b>Sent:</b> Sunday, 11 November 2007 5:19 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] RTP traffic not being forwarded<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal>Hi Ryan,<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Are the SIP and RTP ports are randomly selected or there are
specific ports for these? Unchecking<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>random port selection option on the device/softphone may
help.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>--Vivek<br>
<br>
<o:p></o:p></p>
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<p class=MsoNormal><span class=gmailquote>On 11/10/07, <b>Ryan Newington</b>
<<a href="mailto:ryan@lithiumblue.com">ryan@lithiumblue.com</a>> wrote:</span>
<o:p></o:p></p>
<p class=MsoNormal>Hi Luki,<br>
<br>
Thanks for your advice. I've checked the firewall and it is set to allow all
incoming traffic. I changed the media port range as well with no success. <br>
<br>
Some calls work fine. This is the configuration that doesn't work. The RTP
traffic passes along the chain fine, but the Asterisk server doesn't do
anything with the packets it gets from the near-end SIP phone and the media
gateway. <br>
<br>
SIP Phone <-> Media Gateway <-> Asterisk <-> SIP Phone<br>
<br>
An asterisk internal call will work fine. Eg;<br>
<br>
SIP Phone <-> Asterisk <-> SIP Phone<br>
<br>
Regards<br>
<br>
Ryan<br>
<br>
<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com
</a>] On Behalf Of Luki<br>
Sent: Sunday, 11 November 2007 12:52 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] RTP traffic not being forwarded<br>
<br>
> When using 'rtp debug' on the asterisk console, it shows that it is <br>
> receiving traffic from one endpoint, but not the other. A wireshark trace<br>
> reveals it is actually receiving traffic from both ends.<br>
<br>
Sounds like a firewall issue. Wireshark shows what's "on the wire", <br>
i.e. before iptables. The packets are being dropped for whatever<br>
reason and never reach the asterisk process. Check your iptables and<br>
RTP port range, and perhaps try changing it.<br>
<br>
Luki<br>
<br>
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<p class=MsoNormal><o:p> </o:p></p>
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