One way or no way audio after resume from call hold

Unanswered Question
Feb 5th, 2010
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My configuration is SPA9000 + SPA400 + SPA942's, all with latest firmware as of 2/4/2010.  Using the SPA400 for voicemail on line 1, line 2 is registered to my ITSP, Nexvortex.


Everything is working fine, except......


Sometimes (not always) when I put an inbound caller on hold (actually, either park or hold), when I resume the call, we lose the audio connection to the inbound caller. I have seen the audio die in one or both directions.   Of course when this happens the remote caller gets frustrated and hangs up.  Note that the call is still connected, one or both parties just cannot hear each other.  Prior to placing the call on hold the audio is fine.


This seems to happen more frequently the longer the caller is on hold, usually after about 30 to 60 seconds.  It is definitely not consistent though.


When the inbound caller is first placed on hold they get the SPA9000 generated music, just fine. If we loose the audio in the outbound direction and we subsequently place them on hold they do not get the music on hold.


Its as if one or both sides have forgotten the address of the other side.....


Any suggestions?  My ITSP support in the past has been very good and I can enlist their help if needed.

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Alberto Montilla Thu, 02/11/2010 - 10:29
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Dear Sir;


Your description looks like UDP ports being closed by the router. While I check with engineering (I had a case like this loooong time ago, so 99% but not 100% sure), can you please try with setting [Line x] NAT Keep Alive Enable: YES. This should enable RTP packets being sent even if the call is on hold, so RTP ports are kept open.


Regards
Alberto

markgruber Thu, 02/11/2010 - 10:40
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Thanks for the response Alberto!  I will try your suggestion.  Your thought makes sense as when we using the SPA400 we did not have the problem, probably because the RTP was not going through the router.


Our router is a Pix 501.  I have not explicitly set it to drop the connections after some period of time.  But I will research that as well.


Mark

markgruber Tue, 03/09/2010 - 07:00
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I have tried the NAT keepalive and that does not work.  Still get no audio in both directions after call on hold is resumed after about 30 seconds.  Seems to be for inbound calls only.


I did try one other experiment.  I set up an Xlite softphone and connected it directly to my ITSP (this is all behind my Cisco PIX router).  I was able to put a call on hold and resume with no lost audio after more than a minute on hold.  But this was for an outbound call only.  I will test inbound with the soft phone as well.


Is it worthwhile to turn on logging/debug at this point?  Will there be useful info collected or do I need to use Wireshark?


Mark

Alberto Montilla Thu, 03/11/2010 - 05:37
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Dear Sir;


We would need both logs and wireshark. This way we should be able to determine where the issue is...


One possibility to try: In the SIP tab, set the RFC2543 hold to NO. reboor and try it. This may be a source of interop issue as well.



Regards
Alberto

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