Issue with SIP audio calls in SIP as RTMP .

[Max]

Good day.
We've checked your server and strongly recommend you to do the following:
1. Update WCS to latest build from this page. Now you have an outdated test version that is not intended for production use
2. Increase server RAM to allocate more Java heap memory for WCS. Now, you have default setting

-Xmx1024M

and GC works every 2-3 second, it means not enough memory for Java heap in your usage scenario
3. Set up all your SIP switches to support telephone-event codec and send it in SIP SDP. Now, when WCS send SIP INVITE to switch, it responds 200 OK with SDP not containing telephone-event. This behaviour is not related to SDP fragmentation, so it is not necessary to adjust WCS codecs setttings.

 

[Muhammad Rizwan]

dear team,
i don't know how i can set my all switches to responds 200 OK with SDP containing telephone-event. i have more then 3 switches in production and DTMF is not working now in any of them. you are making things really though for me.
i just don't understand 1 thing if my switches are not responds 200 OK with SDP not containing telephone-event then how come DTMF started working on last upgrade i did on flashphoner server? then it stopped working again.
help me any other work around i am now really getting frustrated.

Thanks

 

[Max]

Good day.
There is a decription of DTMF sending methods supported by WCS, with Asterisk PBX configuration examples, see details here. Another PBX should be configured similarly.
Also please update to the latest WCS build from this page and increase server RAM to allocate more Java heap memory for WCS.