WCS3+Asterisk+EC2 No Audio issue

I

Indrajith K L

New Member
We were using Flashphoner WCS3 + Asterisk + EC2 server for our application. But recently we are having an issue with audio. To check whether the issue is with WCS version we had upgraded to the latest version 3.0.1312. But still having the same issue. We had no issue since the last week though.

Thanks.
 
Last edited:
Max

Max

Administrator
Staff member
First of all make sure UDP ports are open for this EC2 instance.
Try to open all incoming and outbound UDP traffic.
If still no audio, please send tcpdump to logs@flashphoner.com
Code: 
tcpdump udp -s 2048 -i any -w log.pcap
use flag '-i any' if asterisk is installed in one box with WCS.
Send us:
  • pcap
  • server_logs/flashphoner.log
  • console output of chrome browser
We will check.
What is your operation system version? Browser version? Maybe browser was upgraded,
 
Max

Max

Administrator
Staff member
please attach log
flashphoner.log.2015-09-08-11
your flashphoner.log is almost empty
 
Max

Max

Administrator
Staff member
Try to set in flashphoner.properties:
Code: 
ip=[amazon instance external IP]
ip_local=[amazon instance local IP]
Then restart WCS server.
 
Max

Max

Administrator
Staff member
Try to add to flashphoner.properties
Code: 
enable_candidate_harvester=false
and restart WCS server
 
I

Indrajith K L

New Member
Audio is working now (Thanks), but another issue came out. Audio session will cut off after 15 seconds. We tried adding "keep_alive.algorithm =INTERNAL" in server.properties (which solved the same problem in previous working setup) but calling keeps cuting off after 15 seconds. Also the sip client (We use Blink) shows a "media stream timeout" after 15 seconds before call cutoff. Any additional configuration to add on flashphoner.

I have attached log files.. please see the attachments (zipped).

Thanks.
 

Attachments

Last edited:
Max

Max

Administrator
Staff member
Hello,
Keep Alive settings are not applicable here. The keep alive settings are used for websocket and Flash connection only.
As I see from logs and dumps, you have a normal call with two-way voice traffic between Asterisk and WCS.
However Blink terminates the call session.
So it seems as Blink issue.
In the pcap log I see unanswered STUN requests from Blink to Asterisk server. Perhaps if you set another STUN server for Blink, it would work.
 
I

Indrajith K L

New Member
Max said:
Hello,
Keep Alive settings are not applicable here. The keep alive settings are used for websocket and Flash connection only.
As I see from logs and dumps, you have a normal call with two-way voice traffic between Asterisk and WCS.
However Blink terminates the call session.
So it seems as Blink issue.
In the pcap log I see unanswered STUN requests from Blink to Asterisk server. Perhaps if you set another STUN server for Blink, it would work.
Click to expand...
OK, I will check that.

Thanks.
 
Alain R. Mady

Alain R. Mady

New Member
Indrajith K L said:
We were using Flashphoner WCS3 + Asterisk + EC2 server for our application. But recently we are having an issue with audio. To check whether the issue is with WCS version we had upgraded to the latest version 3.0.1312. But still having the same issue. We had no issue since the last week though.

Thanks.
Click to expand...
We have had the same issue and we resolved it easily... Reinstall the Flashphoner server... Good luck
 
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