<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<meta name=Generator content="Microsoft Word 12 (filtered medium)">
<style>
<!--
/* Font Definitions */
@font-face
{font-family:"Cambria Math";
panose-1:2 4 5 3 5 4 6 3 2 4;}
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0cm;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
p
{mso-style-priority:99;
mso-margin-top-alt:auto;
margin-right:0cm;
mso-margin-bottom-alt:auto;
margin-left:0cm;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
span.gmailquote
{mso-style-name:gmail_quote;}
span.e
{mso-style-name:e;}
span.EmailStyle20
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;}
@page Section1
{size:612.0pt 792.0pt;
margin:72.0pt 72.0pt 72.0pt 72.0pt;}
div.Section1
{page:Section1;}
-->
</style>
<!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=EN-US link=blue vlink=purple>
<div class=Section1>
<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'>Thanks for the link. I had a look through and couldn’t find
anything that worked. There are no NAT problems as this is all taking place on
my internal network. The rtp.conf is used to configure the ports. There are no
firewalls or gateways in between these devices.<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'>Asterisk is listening on the correct ports, and receiving the
traffic, as no ICMP messages are being generated to say that the packets could
not be delivered.<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>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<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> Monday, 12 November 2007 5:38 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>well i think rtp port range is defined in rtp.conf and correct
me if i am wrong, these ports must be opened/forwarded to communicate.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><a
href="http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html">http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html</a><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Let me know if you need more information.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Thanks,<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Vivek<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><br>
<br>
<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span class=gmailquote>On 11/11/07, <b>Ryan Newington</b>
<<a href="mailto:ryan@lithiumblue.com">ryan@lithiumblue.com</a>> wrote:</span>
<o:p></o:p></p>
<div>
<div>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Hi Vivek,</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>I'm not sure what you mean,
could you explain further?</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Regards</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Ryan</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:
10.0pt'> <a href="mailto:asterisk-users-bounces@lists.digium.com"
target="_blank">asterisk-users-bounces@lists.digium.com </a>[mailto:<a
href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b>On Behalf Of </b>Vivek Shrivastava <br>
<b>Sent:</b> Monday, 12 November 2007 1:21 AM <o:p></o:p></span></p>
<div>
<p class=MsoNormal><span style='font-size:10.0pt'><br>
<span class=e><b>To:</b> Asterisk Users Mailing List - Non-Commercial
Discussion</span><br>
<span class=e><b>Subject:</b> Re: [asterisk-users] RTP traffic not being
forwarded</span><o:p></o:p></span></p>
</div>
</div>
<div>
<p> <o:p></o:p></p>
<div>
<p>Hi Ryan,<o:p></o:p></p>
</div>
<div>
<p> <o:p></o:p></p>
</div>
<div>
<p>I was just wondering if they need to be according rtp.conf. ( or you may
need to modify rtp.conf)<o:p></o:p></p>
</div>
<div>
<p> <o:p></o:p></p>
</div>
<div>
<p>Regards,<o:p></o:p></p>
</div>
<div>
<p> <o:p></o:p></p>
</div>
<div>
<p>Vivek<br>
<br>
<o:p></o:p></p>
</div>
<div>
<p>On 11/11/07, <b>Ryan Newington</b> <<a href="mailto:ryan@lithiumblue.com"
target="_blank">ryan@lithiumblue.com</a>> wrote: <o:p></o:p></p>
<div>
<div>
<p><span style='font-size:11.0pt;color:#1F497D'>Hi Vivek,</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;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.</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>I can change the port range on
the media server, asterisk and the device, but neither seems to help.</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>My diagram below is probably
misleading. The RTP traffic flow that I see is as follows (one way traffic into
Asterisk)</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>SIP Phone <---> Media
Gateway <b>---></b> Asterisk <b><---</b> SIP Phone</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Ryan</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:
10.0pt'> <a href="mailto:asterisk-users-bounces@lists.digium.com"
target="_blank">asterisk-users-bounces@lists.digium.com </a>[mailto:<a
href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b>On Behalf Of </b>Vivek Shrivastava <br>
<b>Sent:</b> Sunday, 11 November 2007 5:19 PM </span><o:p></o:p></p>
<div>
<p><span style='font-size:10.0pt'><br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] RTP traffic not being forwarded</span><o:p></o:p></p>
</div>
</div>
<div>
<p> <o:p></o:p></p>
<div>
<p>Hi Ryan,<o:p></o:p></p>
</div>
<div>
<p> <o:p></o:p></p>
</div>
<div>
<p>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>random port selection option on the device/softphone may help.<o:p></o:p></p>
</div>
<div>
<p> <o:p></o:p></p>
</div>
<div>
<p>--Vivek<br>
<br>
<o:p></o:p></p>
</div>
<div>
<p>On 11/10/07, <b>Ryan Newington</b> <<a href="mailto:ryan@lithiumblue.com"
target="_blank">ryan@lithiumblue.com</a>> wrote: <o:p></o:p></p>
<p>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" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto: <a href="mailto:asterisk-users-bounces@lists.digium.com"
target="_blank">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>
_______________________________________________ <br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/"
target="_blank">http://www.api-digital.com--</a><br>
<br>
asterisk-users mailing list <br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users"
target="_blank"> http://lists.digium.com/mailman/listinfo/asterisk-users </a><br>
<br>
_______________________________________________<br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/"
target="_blank">http://www.api-digital.com-- </a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users"
target="_blank"> http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p>
</div>
<p> <o:p></o:p></p>
</div>
</div>
</div>
<p><br>
_______________________________________________<br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/"
target="_blank">http://www.api-digital.com-- </a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users"
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p>
</div>
<p> <o:p></o:p></p>
</div>
</div>
</div>
<p class=MsoNormal><br>
_______________________________________________<br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/"
target="_blank">http://www.api-digital.com--</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users"
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
</body>
</html>