[cisco-voip] One-way audio issues with Cisco SIP phones

[Michael Hamann]

Hello,

just to follow up with the one-way audio issue: it seems that this is a software
bug in CUCM 9.1.1

CSCug50634
  :<https://tools.cisco.com/bugsearch/bug/CSCug50634/?reffering_site=dumpcr>

one-way voice with diverted 3rd party phones, from sip via mgcp/qsig

We will upgrade the cluster next weekend. Thanks go out to cisco tac !

Michael

> "Ryan Ratliff (rratliff)" <rratliff at cisco.com> hat am 16. Januar 2014 um 15:53
> geschrieben:
>
>
> That's also a rather old phone firmware version for a new deployment.
>
> You can also look at the phone's web page under Streaming Statistics to see
> the IP address and port it is sending to. Compare with the original call and
> then after hold/resume to see what's different.
>
> If it is a UCM bug you'll have to go to 9.1(2) to get any fixes so I'd
> recommend sticking that on your list of things to do.
>
> -Ryan
>
> On Jan 15, 2014, at 9:02 AM, Mark Holloway <mh at markholloway.com> wrote:
>
> Sounds like the 9971 doesn’t know to send media to the 2901 until hold/resume
> is pressed and then the SIP SDP has the correct address/port to send media to
> the 2901.Apparently receive is working. I would do a debug or wireshark
> capture and see if there is a difference in the SDP when the call is initially
> placed vs. hold/resume.
>
>
> On Jan 15, 2014, at 2:11 AM, Michael Hamann <mail at mhamann.net> wrote:
>
> > Hello,
> >
> > we have a strange issue with one-way-audio under certain circumstances with
> > Cisco SIP phones (8961,9951,9971) registered to a CUCM (9.1.1.20000-5).
> >
> > We are in progress of a migration project from an old Siemens PBX to Cisco
> > UC. At the moment the Cisco UC world is behind the Siemens PBX connected
> > with a Cisco 2901 Voice gateway connected via E1/QSIG. All calls to the PSTN
> > are routed through the Siemens PBX.
> >
> > So calls go like this:
> >
> > CP9971 ---SIP--- CUCM ---MGCP--- Cisco2901(E1) --- QSIG/ISO --- Siemens
> > Hipath4K --- SiemensUP0Phone ---> forwarded to another extension (internal
> > or external)
> >
> > At the moment we have about 350 phones running on the Cisco UC side. So far
> > we used only SCCP phones (79XX phones and 6945 phones). With these phones
> > everything is working fine. No problems with audio or any feature like
> > transfer, conference etc.
> >
> > The SIP phones can do normal calls without problem and a reachable like
> > expected. Call transfer, call hold, conference everything works fine. But
> > magic happens when we try to make a call from a Cisco *SIP* phone to one of
> > the old Siemens extension which has been forwarded to another extension.
> >
> > In this case we get one way audio (the called person can´t hear me). This is
> > reproducible and happens every time. When we forward another cisco extension
> > to something and call this number - no problem.
> >
> > We did some tests with this issue and found out the following:
> >
> > - if we put this "one-way-audio" call on the SIP phone on hold for a moment
> > and get the call back, the one-way-audio is gone and both audio directions
> > work.
> > - this issue only happens on SIP phones. When we put the same DN as a
> > shared/or single line on a SCCP phone, we have no audio issues at all.
> > - calls to forwarded Cisco extension work without audio issues.
> > - for tests we enabled the "media termination point required" option under
> > the SIP phone, now, the one way audio issues are gone, but attended transfer
> > is not possible anymore.
> > - we can´t see any error messages on the phone logs.
> > - when we show up the call status (while the call is established) on the
> > 99XX phone, the packet counters count up in both receiving and sending
> > direction even if the called person can´t hear me.
> >
> > Here some detailed information about software version we use:
> >
> > Cisco SIP phones (8961,9951,9971) --- 9-2-3-27
> > Cisco Voice Gateway 2901 --- IOS 15.2(3)T
> > CUCM --- 9.1.1.20000-5
> >
> > There is no Firewall/NAT between the Phone, CUCM and gateway.
> >
> > We are a bit stuck with this at the moment. Do you have any advice on where
> > to start troubleshooting? Is there any known bug which leads to this issue?
> >
> > thank you for any help
> > kind regards
> > Michael
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>
>
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