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Failing sip Call

Posted on 2014-01-08
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Last Modified: 2014-01-25
Say,

We are making a sip to sip call from one sip instrument to another sip instrument via a hosted server on a fixed IP address.

We find that intermittently the calls between the 2 soft phones will drop at about 19-20 seconds - on the other hand on some occasions the call will last its full duration without any problems.

We are using codec G.729.

Can you suggest what could be causing this issue & a method on how we can solve it?

TX
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Question by:shaunwingin
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12 Comments
 
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Expert Comment

by:José Méndez
ID: 39765690
Try to always mention the type of equipment you are using, otherwise you will have us shooting bullets in plain darkness.

Is that an Asterisk hosted server? What type of SIP phone  are we talking about here?
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Expert Comment

by:José Méndez
ID: 39765695
When you say drop, how does that translate into user experience? Do callers get busy tone? Do they see the call connected with no audio? Do they just see the phone back on-hook?
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Author Comment

by:shaunwingin
ID: 39766104
Sorry for being so concise:
Asterisk PBX is on fix ip and hosted.
Snom 300 calling a Siemens phone.
User experience is that call just goes dead - ie. no sound - They see the call connected with no audio

Pls let me know if need more info.
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LVL 20

Expert Comment

by:José Méndez
ID: 39766422
If the call does not drop, meaning the 2 phones show it as ongoing, my suspicion is that one of these happens:

The 2 phones stop sending audio due to buggy behavior, creating a dead air situation (not likely)

The phone system signals them to stop the audio stream (not likely)

The routing of the audio stream breaks (very likely)

To find out, you will have to collect a packet capture from the 2 IP Phones simultaneously in order to determine when does the audio stream breaks.
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Expert Comment

by:José Méndez
ID: 39766483
Are you 100% sure these 2 phones reside within the same local area network?
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Author Comment

by:shaunwingin
ID: 39767248
It makes sense to me too that
"The routing of the audio stream breaks (very likely)"

The phones are on ADSL lines and hence we are reliant on the cloud.

Why would it only drop just short of 20seconds reliably??

Would a wireshark collection on the hosted server assist with seeing whats going on with the phones while issue occurs? I can perhaps send this to you for analysis your email due to ip security concerns ?
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Author Comment

by:shaunwingin
ID: 39770335
Say any ideas - anyone?
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Accepted Solution

by:
José Méndez earned 500 total points
ID: 39771471
The reason why it lasts 20 seconds is unknown to me. Now, if the 2 phones involved in our scenario are apart from each other´s LAN, linked by an ADSL connection,..... that where I would probably focus on.

The packet capture out of the hosted server is not going to be helpful if the audio path is negotiated between endpoints. Meaning, if the audio flows from 1 phone over to the other directly, the server has no part in handling it. Now it may be possible that due to NAT issues the server is involved.

Can you install 2 softphones instead of the desk phones and replicate the issue? If we were to use softphones we could easily run Wireshark on the PCs and dissect the SIP signaling.
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Author Comment

by:shaunwingin
ID: 39774424
Tx for feedback - yes I can certainly install softphones and run wireshark - how shall we then analyse without sending data on public forum?
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Expert Comment

by:José Méndez
ID: 39774881
Well, you may want to post a disposable mail address where I can reach you, and then you can reply back with the captures.
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Author Comment

by:shaunwingin
ID: 39775821
Tx - moretolife365

This is a gmail adress

Please notify me when sent email so I can check it.
Tx
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Expert Comment

by:José Méndez
ID: 39776385
done.
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