[cisco-voip] SIP trunk one way audio
Buchanan, James
jbuchanan at presidio.com
Fri Nov 12 12:41:04 EST 2010
While I agree with you, if it needs one for transfer or dtmf, it will not allocate one mid-call.
James Buchanan | Technology Manager, UC | South Region | Presidio Networked Solutions
12 Cadillac Dr, Suite 130, Brentwood, TN 37027 | jbuchanan at presidio.com<mailto:jbuchanan at ctiusa.com>
D: 615-866-5729 | F: 615-866-5781 | www.presidio.com<http://www.presidio.com/>
CCIE #25863, Voice
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley
Sent: Friday, November 12, 2010 11:29 AM
To: 'Mark Holloway'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP trunk one way audio
Your right, but I don't need to have the check box to require one on the trunk. It should allocate one from the MRGL on an as needed basis.
From: Mark Holloway [mailto:mh at markholloway.com]
Sent: Friday, November 12, 2010 11:28 AM
To: Bill Riley
Cc: 'Ryan Ratliff'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP trunk one way audio
You should be using an MTP for your SIP trunk to support DTMF. It does not need to be a hardware MTP resource.
On Nov 12, 2010, at 10:05 AM, Bill Riley wrote:
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.
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Friday, November 12, 2010 9:00 AM
To: Bill Riley
Cc: 'Cheng, Karen'; cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] SIP trunk one way audio
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.
If no MTP is being used for the call, try forcing it to use one and see if that fixes the issue.
If you are using media flow-through (default) does changing it to flow-around fix the issue?
-Ryan
On Nov 12, 2010, at 8:09 AM, Bill Riley wrote:
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.
From: Cheng, Karen [mailto:Karen.Cheng at racq.com.au]
Sent: Thursday, November 11, 2010 9:14 PM
To: 'Bill Riley'
Subject: RE: [cisco-voip] SIP trunk one way audio
Not sure if you have checked already but is your SIP trunk using one interface and your MTP/SCCP interface a different interface?
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.
Regards
Karen Cheng
Voice Network Engineer
From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley
Sent: Friday, 12 November 2010 2:15 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] SIP trunk one way audio
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.
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