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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>When I change it to MTP required on the sip trunk everything works as expected. The point is that I shouldn’t need to have this configuration. <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><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><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"'> Ryan Ratliff [mailto:rratliff@cisco.com] <br><b>Sent:</b> Friday, November 12, 2010 9:00 AM<br><b>To:</b> Bill Riley<br><b>Cc:</b> 'Cheng, Karen'; cisco-voip@puck.nether.net<br><b>Subject:</b> Re: [cisco-voip] SIP trunk one way audio<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I see lots of discussion around config and call flow but have you actually done any troubleshooting? One way audio most of the time comes down to the simple fact that one party is not receiving RTP from the other. For these one-way audio calls you need to determine what IP addresses are involved. Next verify this in the signaling via the SDPs in the SIP messages. You can then use show commands on the router to confirm where it thinks it should be sending and receiving RTP to/from and if in fact packet counters are incrementing.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>If no MTP is being used for the call, try forcing it to use one and see if that fixes the issue.<o:p></o:p></p></div><div><p class=MsoNormal>If you are using media flow-through (default) does changing it to flow-around fix the issue? <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif";color:black'>-Ryan<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>On Nov 12, 2010, at 8:09 AM, Bill Riley wrote:<o:p></o:p></p></div><p class=MsoNormal><br><br><span class=apple-style-span><span style='font-size:13.5pt;font-family:"Lucida Grande","serif"'><o:p></o:p></span></span></p><div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I shouldn’t need an MTP for this connection. All SIP traffic is sourced from one interface. I do have SCCP traffic sourced from a different interface but it is only used for a conference bridge, not MTP.</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in;border-width:initial;border-color:initial'><div><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span class=apple-converted-space><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Cheng, Karen [mailto:Karen.Cheng@racq.com.au]<span class=apple-converted-space> </span><br><b>Sent:</b><span class=apple-converted-space> </span>Thursday, November 11, 2010 9:14 PM<br><b>To:</b><span class=apple-converted-space> </span>'Bill Riley'<br><b>Subject:</b><span class=apple-converted-space> </span>RE: [cisco-voip] SIP trunk one way audio</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div></div></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> <o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Not sure if you have checked already but is your SIP trunk using one interface and your MTP/SCCP interface a different interface?</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I had one-way audio/no audio problems ages back due to this because our integrator had configured the SIP trunk to point to int gi0/0’s IP and then configured the SCCP interface to loopback0.</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Regards</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><div><div><p class=MsoNormal><b><i><span lang=EN-AU style='font-size:10.0pt;font-family:"Arial","sans-serif";color:#003478'>Karen Cheng</span></i></b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div></div><div><div><p class=MsoNormal><i><span lang=EN-AU style='font-size:10.0pt;font-family:"Arial","sans-serif";color:#003478'>Voice Network Engineer</span></i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div></div><div><div><p class=MsoNormal><span lang=EN-AU style='color:#1F497D'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div></div><div><div><div><p class=MsoNormal><span lang=EN-AU style='color:#1F497D'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div></div></div></div><div><p class=MsoNormal><span lang=EN-AU style='color:#1F497D'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in;border-width:initial;border-color:initial'><div><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span class=apple-converted-space><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><span class=apple-converted-space> </span>[mailto:cisco-voip-bounces@puck.nether.net]<span class=apple-converted-space> </span><b>On Behalf Of<span class=apple-converted-space> </span></b>Bill Riley<br><b>Sent:</b><span class=apple-converted-space> </span>Friday, 12 November 2010 2:15 AM<br><b>To:</b><span class=apple-converted-space> </span><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><b>Subject:</b><span class=apple-converted-space> </span>[cisco-voip] SIP trunk one way audio</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div></div></div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>I have a new SIP trunk terminating in a 2921 CUBE bundle. When I call in from the Trunk directly to an IP phone it works correctly. If I call from the trunk to IP phone and the IP phone transfers the call without waiting for the remote party to answer I get one way audio. From reading this looks like and MTP issue but I have an MTP set in the MRGL for the trunk.<o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-AU><br></span><span lang=EN-AU style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:green'>75% of inspected vehicle have a defect*. Call 13 1905 and book an RACQ Vehicle Inspection today. *Based on the RACQ Vehicle Defect Report January 2008<br><br></span><span lang=EN-AU><br></span><span lang=EN-AU style='font-size:10.0pt;font-family:"Arial","sans-serif";color:gray'>Please Note: If you are not the intended recipient, please delete this email as its use is prohibited. RACQ does not warrant or represent that this email is free from viruses or defects. If you do not wish to receive any further commercial electronic messages from RACQ please e-mail<span class=apple-converted-space> </span><a href="mailto:unsubscribe@racq.com.au">unsubscribe@racq.com.au</a><span class=apple-converted-space> </span>or contact RACQ on 13 19 05.</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><p class=MsoNormal><span style='font-size:13.5pt;font-family:"Lucida Grande","serif"'>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>