Audio Dropping but Call Stays Connected - Briefly

[Bassackwards]

Hi All,

I don't recall exactly when this issue started happening ( its been months now ) but its driving everyone crazy. Here is the scenario.

1. Both Incoming & outgoing calls
2. Voice/Sound drops at 10 minutes & ~ 35-40 seconds. In all of my testing yesterday, this happens 90-100% of the time.

The voice drops out at a specific time and doesn’t return. The call typically wont terminate during the next few minutes but still shows connected ( the phone call timer even continues ). The user hangs up because there is no voice and the wireshark captures ( taken on the PBX ) shows that the user hung ( BYE ) up but I don’t see ( maybe I don’t know what I’m looking for ) anything that relates to the voice dropping out.

I’ve been working with Speakeasy ( our SIP Trunk provider ). After 3 weeks of looking over the issue and submitting umpteen samples, this is what they returned to me today.

Response from Speakeasy's network operations team regarding my captures ( Wireshark, Edgemark, and Speakeasy's Core Datacenter ).

" I'm afraid that your PBX is actually changing the port that it is sending audio through after we refresh the call with a reinvite, this appears to be the cause of the issues that we are seeing."

UPDATE ADDITION TO Speakeasy TICKET

I can provide a copy of the SIP signaling between that is causing the issue, I might also be able to get you a copy of the RFC that references the Re-INVITE process to help get some attention on that.

The basics of the issue are as follows:
- Speakeasy refreshes the call every 10 minutes. In your situation we do so with a re-INVITE using all the same parameters.
--> Your system appears to be restarting audio on a new port when it receives this, instead of replying with the original port

For example in the example from 10:24 on the 30th, your call starts with audio on port 17120, but refreshes at on port 17124 cutting off the outbound audio.


So, I'm at a loss. Anyone have any idea's? If you want my network architecture, please let me know and I will post it.

Thanks

Brian

 

[SY]

To avoid miscommunication and clarify situation.
From the one hand, PBX does not alter media connection attributes in specified case. It is for sure.
From the other hand, no reasons to ignore information from Voip provider.
So, It is very probable that the problem resides somewhere in the functionality of NAT/firewall which is used to communicate with VoIP provider.

Unfortunatelly, we don't have any information from the PBX...

 

[Bassackwards]

Thank You SY for your response.

I take it that your saying that the PBX doesn't alter the media connection but that we shouldnt discount the VOIP provider. I'm trying to get a copy of the SIP signalling from them. Once I have it, I will post it and maybe someone will see something in there that make sense.

For reference, this is a diagram of my network and how the PBX is connected.
http://dl.dropbox.com/u/11456254/DB%20Design%20v3.jpg

I'll see if I can dig up some wireshark captures.


Thanks again SY.

 

[SY]

Meantime, please post a screenshot of the "Settings->Network" page from "Management console"
Thanks
P.S. It is not necessary to post a link to a third party server. you can attach picture to your post using "upload attachment" at the bottom of "edit message" form.

 

[Bassackwards]

Here you go.

Would you like STUN Server tab, Firewall tab, etc?

Thanks

-Brian

 

[SY]

Here you go.

Would you like STUN Server tab, Firewall tab, etc?

Thanks

-Brian

No, I don't need it.
You have posted following quotation of the answer received from your VoIP provider

For example in the example from 10:24 on the 30th, your call starts with audio on port 17120, but refreshes at on port 17124 cutting off the outbound audio.

I've suspected that the problem is:

So, It is very probable that the problem resides somewhere in the functionality of NAT/firewall which is used to communicate with VoIP provider.

Thanks for the screenshot, it demostrates that I was right.

 

[Bassackwards]

Thank you for your help. I greatly appreciate it .

If/when there is further development on this topic, I'll post an update..

-Brian