[cisco-voip] one way audio after SIP cupover

[Ryan Huff]

The key here, I believe, “another odd fact is that, one of the problem area, for the operator there, she has been able to retrieve the call by placing the call on hold then resuming and the RTP stream seams to establish”

Enable “MTP required” on the CUCM SIP trunk (and verify the SIP trunk has access to a MRGL/MRG that uses a CUCM node as an MTP) and then test. My strawman’s guess is it makes the issue go away.

Let us know the results if you’re able to try this.

Sent from my iPhone

On Feb 5, 2018, at 6:27 PM, Jonatan Quezada <jonatan.quezada at chemeketa.edu<mailto:jonatan.quezada at chemeketa.edu>> wrote:

The crazy part is that this is only happening in 2 of the buildings on main campus. I have 6 other out reaches connecting back here over a metro ethernet infrastructure, and all of them are not any issues.

another odd fact is that, one of the problem area, for the operator there, she has been able to retrieve the call by placing the call on hold then resuming and the RTP stream seams to establish. I followed up with her today and shes said every time she can get two way audio to establish if it seems to choke at first.



On Mon, Feb 5, 2018 at 11:36 AM, Anthony Holloway <avholloway+cisco-voip at gmail.com<mailto:avholloway+cisco-voip at gmail.com>> wrote:
The fact that you received 2 packets is interesting.  Tells me that there is routing happening correctly...to some degree.

If you go to the web page of the phone and click on stream 1, does the far end IP address match your CUBE address?

Also, there's a lot of settings that need to be considered when implementing SIP, such as:

Early Offer and MTP usage
PRACK/Early Media
Offfer/Answer (Capabilities)
Interface Binding
Transport Protocol
OPTIONS Ping
Duplex Streaming
Midcall Signaling
Timers
etc.

Depending on your setting, a lot of different possibilities exist for why you might have the experience you have.  If you could paint a clearer picture of your scenario, that might help out.

On Fri, Feb 2, 2018 at 5:47 PM Jonatan Quezada <jonatan.quezada at chemeketa.edu<mailto:jonatan.quezada at chemeketa.edu>> wrote:
I get that this is usually routing but, is it also routing when the issue is intermittent?

our call flow is like so

CentLink(Provider) ----siptrunk30Meg-PPP(IQ-private)---Cube---CUCM10.5, uccx,unity

<image.png>

bonus facts, I have an operator who is in one of the two most affected buildings and she can recover the call after hold, resume,hold,resume sequence. then full rtp stream is there and she can hear and speak with caller.

are there SIP state change timers I can adjust, I want to tread lightly though because out of all of our outreachs seperated by a metro ethernet hub and spoke topology and almost 30 buildings here on main campus only 2 seem to be affected.





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For immediate assistance please reach out to Chemeketa IT Help Desk at 5033997899<tel:(503)%20399-7899>
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Johnny Q
Voice Technology Analyst - TelNet
Chemeketa Community College
Johnny.Q at chemeketa.edu<mailto:Johnny.Q at chemeketa.edu>
Building 22 Room 131
Work 5033995294<tel:(503)%20399-5294>
Mobile 9712182110<tel:(971)%20218-2110>
SIP 5035406686<tel:(503)%20540-6686>
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For immediate assistance please reach out to Chemeketa IT Help Desk at 5033997899
-or-
Visit the help center from your employee dashboard found here:
https://dashboard.chemeketa.edu/helpcenter/default.aspx


Johnny Q
Voice Technology Analyst - TelNet
Chemeketa Community College
Johnny.Q at chemeketa.edu<mailto:Johnny.Q at chemeketa.edu>
Building 22 Room 131
Work 5033995294
Mobile 9712182110
SIP 5035406686
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip
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