[cisco-voip] Slow to connect calls
Cristobal Priego
cristobalpriego at gmail.com
Thu Jun 25 14:51:53 EDT 2009
Jeff,
I'd do what Wes is saying, however I will check the traces myself before I
send those to TAC.
Trace the call look for the StationOpenReceiveChannel output, this contains
the media Payload and bit rate check the timestamp of this message, then
look for this message StationOpenReceiveChannelAck and then look for
StartMediaTransmission, this message commands the phone to start streaming
RTPs, this includes UDP port, IP of remote endpoint, packet size and codec.
check the timestamps and if the delay between those messages is greater than
40ms or 200ms I don't remember hopefully someone knows the roundtrip delay
for MGCP. then the problem is MGCP and you may want to consider H.323.
I have an entire call flow from a few tests calls I made and got all the
messages that are involved
Call Processing Behavior
P->CM: OffHookMessage
CM->P: StationCallState: OffHook (1)
CM->P: StationDisplayPromptStatus: update the display
CM->P: StationSelectSoftKeys: loads the appropiate soft key set, depending
on the call state
CM->P: StationActivateCallPlane: contains the specified line appearance of
the DN being called
CM->P: StationStartTone: InsideDialTone, commands the phone to play
dialtone
P->CM: KeyPadButton: keypad button was pressed. *= 0xe; #=0xf
CM->P: StationStopTone: msg sent when a tone needs to be stopped (i.e.
DialTone)
CM->P2: StationCallState: State of the call: OffHook=1, OnHook=2,
RingOut=3, RingIn=4, Connected=5
Busy=6, Congestion=7, Hold=8, CallWaiting=9,
CallTransfer=10, CallPark=11, Proceed=12
CallRemoteMultiline=13, InvalidNumber=14
CM->P: StationStartTone: (outsidedialtone=34)
P->CM: keyPadButton:
CM->P: StationStopTone
CM->P2: StationCallInfo: msg has the called party DN/Name and Calling
party DN/Name
CM->P2: StationSetRinger: sets the ringer to the specified ringing mode:
StationRingOff: stops ringer from Ringing, StationInsideRinging: indicates
OnNetCall,
StationOutsideRing: indicates OffNetCall,
StationFeatureRing: used by third-party apps to invoke special features
CM->P2: StationDisplayNotify: this msg causes the phone to discard msg txt
from StationOutputDispalyText and play the text contained in
StationDisplayNotify
CM->P2: StationDisplayPromptStatus
CM->P2: StationSelectSoftKeys
CM->P: StationCallState: proceed=12
CM->P: StationCallInfo
CM->P: StationStartTone: alerting Tone, ringback tone
CM->P: StationCallState: ringout=3
CM->P: StationSelectSoftKeys
CM->P: StationDisplayPromptStatus
P2->CM: OffHookMessage
CM->P2: StationClearNotify: msg sent to the phone to clear the information
sent in the StationDisplayNotify msg
CM->P2: StationSetRinger: RingerOff
CM->P2: StationCallState
CM->P2: StationActivateCallPlane
CM->P: StationStopTone
CM->P: StationCallState
CM->P: StationCallInfo
CM->P: StationSelectSoftkeys
CM->P: StationDisplayPromptStatus
CM->P: StationOpenReceiveChannel: contains the media Payload and bit rate,
asks the phone if it is ready to receive RTP Stream
CM->P2: StationOpenReceiveChannel
P->CM: StationOpenReceiveChannelAck
P2->CM: StationOpenReceiveChannelAck
CM->P2: StartMediaTransmission: commands the phone to start streaming RTP.
Includes: UDP port , IP of remote endpoint, packet size, Codec
CM->P: StartMediaTransmission
P2->CM: OnHookMessage
CM->P2: StationConnectionStatisticsReq: requests connectionstatistics from
the ip phone
P2->CM: StationSetSpeakerMode: turns the speakerphone on/off
CM->P2: StationClearStatus:
CM->P2: StationCallState: 2=onhook
CM->P2: StationSelectSoftkeys
CM->P2: StationDisplayPromptStatus
CM->P2: StationActivateCallPlane
P2->CM: StationConnectionStatisticsRes
CM->P2: StationDefineTimeDate
CM->P2: StationStopTone
CM->P: StationCloseReceiveChannel: commands the phone to stop processing
RTP messages sent to it
CM->P: StationStopMediaTransmission: tells the phone to stop streaming RTP
packets
CM->P2: StationCloseReceiveChannel
CM->P2: StationStopMediaTransmission
CM is The communications Manager
P is phone 1 the phone that is placing the call
P2 is the phone that will receive the call
hopefully this will help you out
Cris
2009/6/25 Wes Sisk <wsisk at cisco.com>
> Unfortunately that is not quite the whole picture. DNA does not
> gracefully identify and handle all overlap conditions which can cause
> delayed routing.
>
> Overlap dial plan is most likely. However, there are other signaling
> issues such as long round trip time that can caused delayed call routing.
> Another cause is hunting through endpoints in a routelist or routegroup
> which are partially responsive.
>
> CCM SDI and SDL traces from the involved CM servers is the best way to
> identify what is introducing the delay. If you are uncomfortable reviewing
> those open a TAC case and attach them. Include on the case:
> traces
> calling party number
> called party number
> approximate time of call based on phone time.
> phone time offset from the server
>
>
> /Wes
>
>
>
> On Thursday, June 25, 2009 2:16:09 PM, Jeff Ruttman
> <ruttmanj at carewisc.org> <ruttmanj at carewisc.org> wrote:
>
> Thanks Cristobal, Ian.
>
> There doesn't appear to be a dial plan problem. See below.
>
> I'm open to changing protocol to H.323, but I wouldn't know how to do that
> exactly at the moment. And anyway as I mentioned we have an H.323 GW for
> each of the sites that use Trunks and POTS. They've always had a status of
> "unknown" and if you view their config, the registration is unknown. 2 of
> the 4 have a device name/IP address that is the same as the routers at those
> sites, and the other 2 have a device name/IP address that is the same as the
> IP on the FXO and FXS CCM configs on the MGCP GWs for these sites. They are
> otherwise configured the same.
>
> Are these H.323 GWs just ornamental?? Are they doing anything? Certainly
> the MGCP GWs are....
>
> I'll keep plugging away. Little by little I'll catch on.
>
> Thanks
> jeff
>
> *Results Summary*
>
> - *Calling Party Information*
> - *Calling Party* = 5801
> - *Partition* =
> - *Device CSS* =
> - *Line CSS* = MO1Phones
> - *AAR Group Name* =
> - *AAR CSS* =
> - *Dialed Digits* = 92977902
> - *Match Result* = RouteThisPattern
> - *Matched Pattern Information*
> - *Pattern* = 9.[2-9]XXXXXX
> - *Partition* = MO1Routes
> - *Time Schedule* =
> - *Called Party Number* = 92977902
> - *Time Zone* = Central Standard/Daylight Time
> - *End Device* = MO1-RL-Local
> - *Call Classification* = OffNet
> - *InterDigit Timeout* = NO
> - *Device Override* = Disabled
> - *Outside Dial Tone* = NO
>
>
> ------------------------------
> *From:* Cristobal Priego [mailto:cristobalpriego at gmail.com<cristobalpriego at gmail.com>]
>
> *Sent:* Thursday, June 25, 2009 11:28 AM
> *To:* Ian MacKinnon
> *Cc:* Jeff Ruttman; cisco-voip
> *Subject:* Re: [cisco-voip] Slow to connect calls
>
> Sounds like Ian is right. you can have a dial plan problem
> do you have a centralized deployment? if you do, you need to be very
> careful with the delay of MGCP, because this is a Master-Slave protocol.
> MGCP has a dependency of callmanager, so before you can place a call, the gw
> needs to talk to the CUCM to know what to do. if that's the case I'd say the
> best option for you is to change your protocol to H.323.
>
>
> Cris
>
> 2009/6/25 Ian MacKinnon <Ian.Mackinnon at lumison.net>
>
>> Hi Jeff,
>>
>> That sounds like a dial plan problem ie it is waiting for another digit,
>> and then timing out.
>>
>>
>>
>> Can you dial the number before hitting dial on the phone so it is all
>> present as opposed to lifting the handset and dialling each digit in turn?
>>
>>
>>
>> *From:* cisco-voip-bounces at puck.nether.net [mailto:
>> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Jeff Ruttman
>> *Sent:* 25 June 2009 14:47
>> *To:* cisco-voip
>> *Subject:* [cisco-voip] Slow to connect calls
>>
>>
>>
>> Greetings,
>>
>>
>>
>> Some of our sites have DID trunk ports and POTS lines, and we have MGCP
>> controlled GWs with FXS and FXO configured. We also have for these sites
>> H.323 GWs--which frankly I'm not sure why or what they do.
>>
>>
>>
>> Anyway, at one of those sites, it takes a count of 15 or more for an
>> outgoing call to connect. I know some delay is expected with that setup,
>> but that's quite a bit longer than at our comparable sites.
>>
>>
>>
>> Is that length of delay still within expectations? Or is there something
>> perhaps I can do to speed that up?
>>
>>
>>
>> Thanks
>>
>> jeff
>>
>>
>>
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