sip problems

[phonedev]

Hello,

We have some sip problems couldn't find any solution since 2 months.
The problems are:
We are using flashphone in our crm. So it is using for mostly outcalls
and in an hour an user can make more than 50 calls.
1- without any reason after 2-3 calls while making a new call outcall
status stucks in pending status and not changing. We tried to solve
this problem from javascript side like this:
If status pending, init flashphoner again -> create new session -> call number
This loop sometimes finalize success call after 1 minute. This is not
acceptable for call center.

2- The voice of our users sometimes not send in good quality to other
side or not send, opposite side says can't hear anything, however our
users can hear them. Our flashphoner server is in region and we get
25-50 ms ping to server.

Can you help us if any solution to fix these main problems,

Regards

 

[Max]

Good day.
Please reproduce the problem and collect a report as described here using report.sh script, then send report archive using this form. If the archive size is more than 30 M, place it on Google Drive and send the link using Comment field in from.

 

[phonedev]

Hello,

I sent the log.pcap and report in one archive file

regards

 

[Max]

We checked the report.
First, to handle 50 calls simultaneously you need a more powerful server, recommended at least 8 CPU / 16 Gb RAM.
The most of SIP calls in logs are established correctly and are finished by caller hangup. The traffic dump was collected after the logs, so we did not see the call from the dump in them, but, according to dump, the call was also established correctly, and RTP traffic was passing between WCS and SIP PBX.
Some calls are not established with the message in log Sip provider address is WRONG!. It means the outbound proxy address cannot be resolved due to some external reasons: DNS failure, network failure etc.
Also there were some tries to register the same SIP login (1003) twice. This should not work, so we recommend to review your flow.
The final recommendations are:
1. Upgrade the server
2. Check the possible network issues leading to DNS failure
3. Review the flow to escape a simultaneous registration of the same SIP login

 

[phonedev]

Hello Max,

We have only 5 users to make calls simultaneously. How it is

 

[Max]

We have only 5 users to make calls simultaneously. How it is

2 CPU / 3 Gb RAM should be enough for 5 streams if there is no audio/video transcoding.

 

[phonedev]

My question deleted some how, it is:
How it is possible 50 calls simultaneously in our server? Is it something wrong?

2 CPU / 3 Gb RAM should be enough for 5 streams if there is no audio/video transcoding.

We make only audio calls, not using video. But I am not sure we have audio transcoding.

 

[Max]

How it is possible 50 calls simultaneously in our server? Is it something wrong?

In the post above, you've mentioned 50 calls, so we checked the server configuration to conform the requirement.

We make only audio calls, not using video. But I am not sure we have audio transcoding.

According to logs and dump, PCMU (G711u) audio codec is used by PBX, so audio transcoding is not needed if Chrome browser is used to make a call and play the SIP stream because Chrome supports PCMU.
So there are two problems you have to solve:
1. Network issues leading to DNS failure
2. Simultaneous registration tries of the same SIP login on PBX (2 REGISTER messages sent subsequentally).