[cisco-voip] SIP trunk one way audio
Bill Riley
bill at hitechconnection.net
Fri Nov 12 12:50:50 EST 2010
I don't think that is correct. It should only need to have one available in
the MRGL, not one allocated every time a call comes in.
voice service voip
ip address trusted list
ipv4 x.x.x.x
ipv4 x.x.x.x
ipv4 x.x.x.x
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0
fallback none
h323
modem passthrough nse codec g711ulaw
sip
bind control source-interface Serial0/0/0:1
bind media source-interface Serial0/0/0:1
early-offer forced
midcall-signaling passthru
!
voice class codec 100
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
dial-peer voice 201 voip
preference 1
destination-pattern 91[2-9]..[2-9]......
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp cs3 signaling
!
dial-peer voice 202 voip
preference 1
destination-pattern 9[2-9]......T
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp ef signaling
!
dial-peer voice 203 voip
preference 1
destination-pattern ^9556[2-9]......
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp ef signaling
!
dial-peer voice 204 voip
preference 1
destination-pattern 9555[2-9]......
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
fax rate 14400
ip qos dscp ef signaling
!
dial-peer voice 411 voip
preference 1
destination-pattern 5555555..
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp cs3 signaling
!
dial-peer voice 412 voip
preference 2
destination-pattern 5555555..
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp cs3 signaling
!
dial-peer voice 413 voip
preference 3
destination-pattern 5555555..
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp cs3 signaling
From: Mark Holloway [mailto:mh at markholloway.com]
Sent: Friday, November 12, 2010 11:32 AM
To: Bill Riley
Cc: 'Ryan Ratliff'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP trunk one way audio
You should have it on the trunk.
On Nov 12, 2010, at 10:29 AM, Bill Riley wrote:
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
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] On Behalf Of Bill Riley
Sent: Friday, 12 November 2010 2:15 AM
To: 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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