asked on Failing sip Call
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
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
Voice Over IPIP TelephonyTelecommunications
Last Comment
José Méndez
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?
ASKER
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.
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.
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.
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.
Are you 100% sure these 2 phones reside within the same local area network?
ASKER
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 ?
"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 ?
ASKER
Say any ideas - anyone?
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ASKER
Tx for feedback - yes I can certainly install softphones and run wireshark - how shall we then analyse without sending data on public forum?
Well, you may want to post a disposable mail address where I can reach you, and then you can reply back with the captures.
ASKER
Tx - moretolife365
This is a gmail adress
Please notify me when sent email so I can check it.
Tx
This is a gmail adress
Please notify me when sent email so I can check it.
Tx
done.
Is that an Asterisk hosted server? What type of SIP phone are we talking about here?