[cisco-voip] MGCP Audio Cut Through Delay
Jonathan Charles
jonvoip at gmail.com
Tue Jul 22 10:50:36 EDT 2008
Ok, the problem is back... This is growing unpleasant... the bugID was
so on the money...
Jonathan
On Fri, Jul 18, 2008 at 12:32 PM, c3voip <c3voip at nc.rr.com> wrote:
> I wish I had that "problem" :)
>
> Only two T-1's to each office for voice and data, but luckily not that many
> calls between offices. Have to rely strictly on QoS policy routing for the
> time being :(
>
> -C
>
> -----Original Message-----
> From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
> Sent: Friday, July 18, 2008 1:07 PM
> To: c3voip; Jonathan Charles; Wes Sisk
> Cc: cisco-voip
> Subject: RE: [cisco-voip] MGCP Audio Cut Through Delay
>
> I never understand why a remote site with 3 7960 Phones with
> MetroEthernet type connectivity (10mbps or higher) would implement
> Locations at 384k or anything other than unlimited. Yes a site with
> 512K Frame Relay I could see locations, but I see customers with 150Mb
> pipes and something like 786k in Locations with 10 phones. Doesn't make
> sense but I see it often.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of c3voip
> Sent: Friday, July 18, 2008 12:48 PM
> To: 'Jonathan Charles'; 'Wes Sisk'
> Cc: 'cisco-voip'
> Subject: Re: [cisco-voip] MGCP Audio Cut Through Delay
>
> I had a problem going from 4.1(3)sr5b to 4.1(3)sr7 with Call Control.
> So
> just a warning make sure your Locations and Regions are working properly
> after upgrading, if you use them.
>
> I'm still waiting for a patch, luckily we have enough bandwidth between
> Locations to allow the unlimited workaround.
>
> -C
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Friday, July 18, 2008 12:35 PM
> To: Wes Sisk
> Cc: cisco-voip
> Subject: Re: [cisco-voip] MGCP Audio Cut Through Delay
>
> So, this turned out to be CSCsc06953...
>
> Patched em to 4.1(3)sr7 and it fixed it.
>
>
>
> Jonathan
>
> On Thu, Jul 10, 2008 at 9:48 AM, Wes Sisk <wsisk at cisco.com> wrote:
>> need to check cm traces to see what happens between answer and audio
> cut
>> through.
>> required CM SDI and SDL traces.
>>
>> /wes
>>
>> Jonathan Charles wrote:
>>>
>>> So, is this a phone firmware issue? The PSTN caller (and the IP
> Phone)
>>> do NOT hear anything for about 4-5 seconds once the IP phone call is
>>> answered.
>>>
>>> This is also inbound only.
>>>
>>> CCM is 4.1(3)sr5b and Gateway is a 2811 running 12.4(3g)
>>>
>>>
>>>
>>>
>>> Jonathan
>>>
>>> On Wed, Jul 9, 2008 at 4:24 PM, Wes Sisk <wsisk at cisco.com> wrote:
>>>
>>>>
>>>> at the end the router reports call statistics:
>>>> P: PS=978, OS=156480, PR=980, OR=156800, PL=0, JI=0, LA=0
>>>> PS = packets sent
>>>> PR = packets received
>>>> 978 is close enough to 980. this means all RTP packets made it
> through
>>>> the
>>>> network between the endpoints.
>>>> 980 packets * 20msec/packet = 19,600msec of audio.
>>>>
>>>> Time from sendrecv to recvonly = 03:51:40.645 - 03:52:00.227 = 19.6
>>>> seconds
>>>> (approximately).
>>>> 19.6 seconds connected and 19.6 seconds of RTP traffic. Thus audio
> was
>>>> successfully established and packets were sent and received
> successfully
>>>> for
>>>> the duration of the connection.
>>>>
>>>> Router responded correctly and offered its ip address to receive RTP
>>>> stream. the mgcp call agent (cm?) sent instruction to establish
>>>> bidirectional audio full 15 seconds later. Need to see what took
> the
>>>> call
>>>> agent(cm?) so long to obtain the media ip/port from the other
> endpoint.
>>>>
>>>> debugs with comments:
>>>> //create connection to setup new call
>>>> Jul 10 03:51:25.534 CST: MGCP Packet received from
> 10.1.1.10:2427--->
>>>> CRCX 89768
>>>>
>>>> //router provides ip & port for RTP stream:
>>>> Jul 10 03:51:25.550 CST: MGCP Packet sent to 10.1.1.10:2427--->
>>>> 200 89768 OK
>>>> I: 6
>>>>
>>>> // router receives instruction to change audio mode to sendrecv.
> this is
>>>> usually 2 way audio
>>>> Jul 10 03:51:40.645 CST: MGCP Packet received from
> 10.1.1.10:2427--->
>>>> MDCX 89773 AALN/S0/SU1/3 at MC003776.mcenery.local MGCP 0.1
>>>> C: A00000000200d51c000000F5
>>>> I: 6
>>>> X: 0
>>>> L: p:20, a:PCMU, s:off, t:b8
>>>> M: sendrecv
>>>> R: L/hu, D/[0-9ABCD*#]
>>>> S:
>>>> Q: process,loop
>>>>
>>>> v=0
>>>> o=- 6 0 IN EPN AALN/S0/SU1/3 at MC003776.mcenery.local
>>>> s=Cisco SDP 0
>>>> t=0 0
>>>> m=audio 18840 RTP/AVP 0
>>>> c=IN IP4 172.20.113.10
>>>> <---
>>>>
>>>> //router receives instruction to stop transmitting audio
>>>> Jul 10 03:52:00.227 CST: MGCP Packet received from
> 10.1.1.10:2427--->
>>>> MDCX 89775 AALN/S0/SU1/3 at MC003776.mcenery.local MGCP 0.1
>>>> C: A00000000200d51c000000F5
>>>> I: 6
>>>> X: 0
>>>> M: recvonly
>>>>
>>>> // router reports call statistics:
>>>> Jul 10 03:52:00.287 CST: MGCP Packet sent to 10.1.1.10:2427--->
>>>> 250 89776 OK
>>>> P: PS=978, OS=156480, PR=980, OR=156800, PL=0, JI=0, LA=0
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> v=0
>>>> c=IN IP4 172.20.115.254
>>>> m=audio 17344 RTP/AVP 0 100
>>>>
>>>>
>>>>
>>>> /Wes
>>>>
>>>>
>>>> Jonathan Charles wrote:
>>>>
>>>> Here you go... no obvious errors...
>>>>
>>>>
>>>>
>>>> Jonathan
>>>>
>>>> On Wed, Jul 9, 2008 at 2:42 PM, Wes Sisk <wsisk at cisco.com> wrote:
>>>>
>>>>
>>>> yep
>>>>
>>>> Jonathan Charles wrote:
>>>>
>>>>
>>>> The MGCP packet debug?
>>>>
>>>>
>>>> J
>>>>
>>>> On Wed, Jul 9, 2008 at 1:02 PM, Wes Sisk <wsisk at cisco.com> wrote:
>>>>
>>>>
>>>>
>>>> can you share the debug (blank out the actual phone numbers) with
> debug
>>>> datetime msec?
>>>>
>>>> /wes
>>>>
>>>> Jonathan Charles wrote:
>>>>
>>>>
>>>>
>>>> We have a remote site via a VPN tunnel (over MPLS), and we are
> seeing
>>>> significant audio cut through delays.
>>>>
>>>> We did a debug mgcp packet and saw no retries.
>>>>
>>>> We can also see the problem when PSTN callers are placed on hold and
>>>> retrieved.
>>>>
>>>> Delay lasts between 3 to 4 seconds...
>>>>
>>>> Any ideas?
>>>>
>>>> MGCP config is the default (no special packages enabled/disabled)...
>>>>
>>>>
>>>>
>>>> Jonathan
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>>
>>>>
>>>>
>>
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