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<body class='hmmessage'><div dir='ltr'>Hi Gonzalo,<br><br>Thanks a lot for your reply. I took a closer look at the issue and I guess we figured out the reason why this is happening. But, still couldnt get the solution unfortunately. Our topology is like this:<br><br>sipml5(behind NAT) -> (Public Interface)kamailio(Private Interface) <----> Freeswitch<br><br>So, when we are sending calls to to freeswitch via Kamailio; Freeswitch is updating the SDP with its private IP. Therefore the client tries to send RTPs to the private IP of freeswitch. We have rtpproxy installed in our Kamailio server. We tried bunch of stuffs to solve the issue like ICE handling with rtpengine but nothing worked out. Can you please give me some more advise that I may try ?<br><br>Thanks in advanced.<br><br><br><br><div><hr id="stopSpelling">Date: Wed, 22 Oct 2014 09:53:58 -0700<br>Subject: Re: [SR-Users] One sided or no voice issue with websockets<br>From: gascagonzalo@gmail.com<br>To: sr-users@lists.sip-router.org<br>CC: dodul@live.com<br><br><div dir="ltr">Hi Kamrul,<input value="{"email-policy":{"state":"closed","expirationUnit":"days","disableCopyPaste":false,"disablePrint":false,"disableForwarding":false,"expires":false},"attachments":{}}" type="hidden"><div><br></div><div>Can you provide more details about the call flow:</div><div>It is important to understand the following:</div><div>-Topology</div><div>-Web browser models and versions (Chrome vs FireFox)</div><div>-Kamailio version</div><div><br></div><div>When you say:</div><div><span style="font-family:arial,sans-serif;font-size:12.8000001907349px;">"It works perfectly within local network"</span><br></div><div><span style="font-family:arial,sans-serif;font-size:12.8000001907349px;">Please take a look at ICE negotiation.</span></div><div><span style="font-family:arial,sans-serif;font-size:12.8000001907349px;"><br></span></div><div>I have sipml5 placing calls to Asterisk via Kamilio working without any issues.</div><div><br></div><div>sipml5 ---> kamailio --> fs --> asterisk</div><div> ------------------ media ------------></div><div><br></div><div>Please provide logs:</div><div>sipml5 (java console)</div><div>kamailio logs</div><div>(If possible browser logs)</div><div><br></div><div>-Gonzalo</div><div><br></div></div><div class="ecxgmail_extra"><br><div class="ecxgmail_quote">On Wed, Oct 22, 2014 at 9:12 AM, Amit Patkar <span dir="ltr"><<a href="mailto:amit@avhan.com" target="_blank">amit@avhan.com</a>></span> wrote:<br><blockquote class="ecxgmail_quote" style="border-left:1px #ccc solid;padding-left:1ex;">Please check ICE server settings. Your browser may be publishing local IP. One way voice or no voice is typical case when client is behind firewall.<br><br>Regards,<br>Amit Patkar<div class="ecxHOEnZb"><div class="h5"><br><br>dodul <<a href="mailto:dodul@live.com" target="_blank">dodul@live.com</a>> wrote:<br><br>
<div>Hi</div>
<div><br>
</div>
<div>I didn't get any responses from anyone regarding my issue. Can someone please give me some clue what I can do? Or if more information is needed please let know so that I can provide. </div>
<div><br>
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<div><br>
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<div>
<div style="font-size:9px;color:#575757;">Sent from my Samsung Galaxy smartphone.</div>
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-------- Original message --------<br>
From: Kamrul Khan <<a href="mailto:dodul@live.com" target="_blank">dodul@live.com</a>> <br>
Date:10-21-2014 17:45 (GMT-06:00) <br>
To: <a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a> <br>
Cc: <br>
Subject: [SR-Users] One sided or no voice issue with websockets <br>
<br>
<div>
<div dir="ltr">Hi,<br>
<br>
We have a setup with sipml5 to kamailio. It works perfectly within local network. In public network the signaling establishes perfectly, but most of the time we hear no voice, sometimes we hear one sided voice and in rare cases we hear voice from both sides.
To fix this issue we configured our nathandler like the below: But, still no luck. Any idea how to fix this? Please HELP!!!<br>
<br>
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")<br>
modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n")<br>
.<br>
.<br>
.<br>
route {<br>
.<br>
.<br>
if (nat_uac_test("115")) {<br>
if(nat_uac_test("64"))<br>
force_rport();<br>
}<br>
if (method=="REGISTER") {<br>
fix_nated_register();<br>
add_rcv_param();<br>
} else {<br>
fix_nated_contact();<br>
if(nat_uac_test("64")){<br>
if (!add_contact_alias()) {<br>
xlog("L_ERR", "Error aliasing contact <$ct>\n");<br>
sl_send_reply("400", "Bad Request");<br>
exit;<br>
}<br>
} else {<br>
add_rcv_param();<br>
}<br>
}<br>
}<br>
.<br>
.<br>
<br>
onreply_route {<br>
if (nat_uac_test(64)) {<br>
add_contact_alias();<br>
}<br>
.<br>
.<br>
}<br>
<br>
<br>
</div>
</div>
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<br></blockquote></div><br></div></div> </div></body>
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