[cisco-voip] Delayed Voice - Troubleshooting Tips?

[Wes Sisk]

for true troubleshooting approach, use detailed CM traces and sniffer 
trace from the back of the phone

gw---cm---p1

for an inbound call from gw thru cm to p1:

1. p1 goes offhook
2. cm receives offhook msg from phone
3. cm sends msg to the gw asking for destination ip:port for the RTP 
stream (OpenLogicalChannel if h323, CRCX if MGCP)
4. cm sends OpenReceiveChannel down to the phone asking for a 
destination ip:port for the RTP stream
5. gw sends port number to cm (OpenLogicalChannelAck if h323, part of 
the 200 OK msg if MGCP)
6. cm sends startmediatransmission to the phone indicating the gw ip:port
7. p1 ARPs for gw IP or default gateway IP based on subnet mask
8. p1 begins transmitting audio to gw
7. p1 sends OpenReceiveChannelAck to cm indicating the ip:port to use 
for the RTP stream
8. cm sends msg to gw with port number for phone OpenLogicalChanelAck if 
h323, MDCX sendrecv if mgcp
9. gw ARPs for phone IP or default gateway IP based on subnet mask
10. gw begins transmitting audio.

In the CM trace check the timing between the offhook message from the 
phone (or answer softkey) to openreceivechannel/openreceivechannelack 
and startmediatransmission.  If delay here, find out why.  If all this 
looks timely, get a sniffer trace of the RTP stream and look for 
anomolies in the stream such as silencesuppression packets, resetting 
sequence numbers, changes in SSRC, duplicate streams to the same port 
number, etc.

But if you're trying 7.1(1), save yourself some time and back out 7.0(2) 
or advance to 7.1(2).
Watch out for the extra button press in 7.1(2) to resume held calls.  
This will be fixed under CSCsa63893 in phone load 7.2(1) so that:
o When there is a single call on hold on a given line, pressing the line 
button will resume it; we had a misunderstanding about this with UI 
architecture due to next bullet.
 o If there is *more* than one call on hold on a given line, pressing 
the line button will toggle between the details view for the line and 
the overview (home plane) view.  To resume a call in this case, the user 
must select the call to be resumed, and then use the "Resume" button.

/Wes



Erick B. wrote:

>The UK end has the 7.1(1) phoneload from the filename
>posted below. That load had numerous voice quality,
>reboot, etc issues and was pulled from cisco. They
>have released a new version out now, 7.1(2) which
>fixes those issues present in 7.1(1). 
>
>Does the delay happen on all calls or just some? Could
>there be a routing issue between the phones perhaps
>when the audio stream is cutting over? Is there a
>firewall, etc between the US and UK that traffic goes
>through? 
>
>--- Anthony Mendoza <amendoza at Niku.com> wrote:
>
>  
>
>>We are experiencing an interesting symptom which I
>>have not been able to
>>identify a fix for.  We have 2 Clusters (UK and US)
>>and when calls are
>>made between the two there is about a 1-3 second
>>delay before the
>>calling party can hear the called party.  Codec
>>being used right now is
>>G.711u and sufficient bandwidth is available.  We
>>didn't have this delay
>>with G.729, but had quality issues when trying to
>>use G.729..  any
>>suggestions on how I should troubleshoot further? 
>>Silence supression on
>>both ends has already been disabled.  The only thing
>>I can think of
>>right now is that the phone load on each end is a
>>different version, but
>>didn't think that should matter.
>>
>>P00307010100 in UK
>>P0030600T017 in US
>>
>>-Anthony
>>
>>
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