Feature #1627
closed
Added by Tim Sudall over 12 years ago.
Updated almost 11 years ago.
Description
On the wiki it mentions that the stun server can be found in the dns srv by the client.
I am a novice user trying to connect to your service from a 3g connection and I have a problem with one way audio. I fear this to be because my client isn't automatically receiving details correctly from the dns srv so I would like to set these manually. is this possible?
Also, should I be using the global IP setting for 3g in my options?
Is there a server address I can try or anything else I can try ?
Files
The only option that can improve things related to media is ICE. Enable ICE and give it a try, if both end-points support ICE, they perform connectivity checks and select a working candidate. ICE needs stun server, but the server is already defined manually as far as I see in your screen shots.
Otherwise, the problem is most likely related to your 3G operator, he may block ports or services on purpose and there is little you can do about it.
Compare the results to calls made on WiFI, if it does not work on 3G the problem is certainly caused by the operator.
- Status changed from New to In progress
- Assignee set to Adrian Georgescu
- Keywords set to nat traversal
As far as in aware my isp doesn't block ports for voip but I'd like to contact them to be sure. what is the Best way to get a definitive answer from them?I know phone provider support is limited, especially regarding data services.
At the moment my setup chain is ipkall DID > sip2sip or ipkall DID > google voice > sip2sip.
Am I right in saying that ice won't work unless the call is between two sip clients both with ice enabled?
Would there be an advantage for me using something like pbexs.com to route my calls or does pbxes also lack ice support ?
thanks for your help.
There is no definitive answer for this. Starting with your microphone, the sip client software and its settings, the selected audio codec, the network packets loss or the SIP server, anything can be the culprit for what you describe. Try to replace any of them to see where is the fault.
Thanks for your reply. Is it possible to answer my two questions previous about using pbexes and ice?
Thanks,I am trying to narrow it down.
I do not know what pbxes is or does.
Okay, thanks. And ICE can only be used between two sip clients if available?
E.g it won't work connected to google voice or a DID such as Ipkall?
- Status changed from In progress to Closed
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