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

[Ryan Ratliff (rratliff)]

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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