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Troubleshooting 1 way audio


DWAyotte
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I have frequent complaints about calls that only 1 party can hear the other. What is the best way to troubleshoot these sorts of issues? I have looked through a packet sniff, but am not seeing anything that looks to be an issue. Perhaps my eyes are not trained to see the proper errors. I am wanting to get some pointers from you guys on how to best troubleshoot these sorts of issues. Thanks a ton.

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is it intermitting?

try on the admin/setting/port page set Lock codec during conversation: to yes

do the same for /admin/domain/selectyourdomain/trunks/ Lock codec during conversation: set it to yes

 

This didn't seem to do the trick. I am still getting a handful of "I can't hear the caller" complaints each day. Any other ideas? I would love to be able to nail this down. Thanks,

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could you provide details on the configuration of the environment?

audio issues are commonly related to NAT routers and routers no properly configured to dynamically manage SIP calls.

On a PBXnSIP server with a Public IP exposed 1-way audio is less troublesome, but can be affected by adjacent devices on public IP's not playing fair..Internally on the LAN 1-way audio is less troublesome but LAN switches not properly configured for 802.1X features can result in the RTP streams being affected.

 

If these calls are just with an external caller, and you are using a SIP provider, then look to any router that you may have in front of PBXnSIP? RTP ports may need forwarding on less smart routers.

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could you provide details on the configuration of the environment?

audio issues are commonly related to NAT routers and routers no properly configured to dynamically manage SIP calls.

On a PBXnSIP server with a Public IP exposed 1-way audio is less troublesome, but can be affected by adjacent devices on public IP's not playing fair..Internally on the LAN 1-way audio is less troublesome but LAN switches not properly configured for 802.1X features can result in the RTP streams being affected.

 

If these calls are just with an external caller, and you are using a SIP provider, then look to any router that you may have in front of PBXnSIP? RTP ports may need forwarding on less smart routers.

 

my pbx sits on my LAN and on the internet, no NAT (2 NICs). I have all the necessary internet facing ports open and the rest closed (a total of 100 RTP ports).

I have had 1 way on LAN more times then I feel comfortable having.

So if it is my switches, what can I do? What 802.1x features should I be verifying?

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my pbx sits on my LAN and on the internet, no NAT (2 NICs). I have all the necessary internet facing ports open and the rest closed (a total of 100 RTP ports).

I have had 1 way on LAN more times then I feel comfortable having.

So if it is my switches, what can I do? What 802.1x features should I be verifying?

How many calls, are you sure 100 rtp ports are enough for you?

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