Author Topic: SIP Call Audio Failure  (Read 1866 times)

Offline ralph

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SIP Call Audio Failure
« on: August 21, 2015, 01:41:12 PM »
Let me bounce this off ya'll.

I have a customer that has just started using SIP for some of their DIDs,
A problem has cropped up where after 10-15 minutes on the call the user stops receiving audio.
The carrier put a capture on their SBC and they can hear audio both directions so it would appear that something is stopping the audio from being sent to the device somewhere between it and the carriers SBC.

Customer says there is no firewalls in between.   I suspect an ALG somewhere but I've no idea where it might be.

Anyone have a suggestion?

Ralph


Offline acejavelin

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Re: SIP Call Audio Failure
« Reply #1 on: August 23, 2015, 09:16:58 AM »
Audio cut-off at 10-15 minutes sounds suspiciously like a Sonicwall Firewall issue... we had this crop up several times.

For a year or so now, any customer with a 3300 or 5000 that gets SIP trunking, we tell them they must have a MBG and a dedicated pipe or it will not work properly (the most common SIP carrier in our area is the cable company, they provide a separate cable modem for this service anyway) . Since then we have had zero SIP issues with any customers with this setup.
« Last Edit: August 23, 2015, 09:18:46 AM by acejavelin »

Offline bluewhite4

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Re: SIP Call Audio Failure
« Reply #2 on: August 23, 2015, 10:42:45 AM »
I suspect an ALG somewhere but I've no idea where it might be.

I would agree with your suspicion.

Without knowing more about the site design, this makes it hard to troubleshoot. Are you able to get a wireshark directly off the 3300 at all?

Offline jonnycreb

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Re: SIP Call Audio Failure
« Reply #3 on: September 10, 2015, 10:28:44 AM »
We are getting similar issues on the BT SIP product - what happens is that the call will connect, then at random times in the call the caller will go silent (downstream audio from SIP). They can hear us talking, but we can't hear them. After about 10 secs or so then they come back in and the conversation can carry on. We have had endless traces carried out at various points on the network from the phone, through core switches, inside and outside firewalls, IPExchange and still can't resolve.

They have identified that the SIP is carrying 2 streams of audio and the MCD doesn't know which one to use. It the SIP provider that sends an additional, 'rogue' RTP stream with a different SSRC (RTP stream identifier) but to the same UDP port number on the Mitel Border Gateway. Because the proxy device is receiving two RTP streams concurrently it does not know which one to proxy to the SIP User Agent (Mitel MN3300) and as such does not proxy either.

Any ideas welcome......


 

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