Hi Friends,
I am trying to connect two SJPhones with SIP server. And i need to redirect the RTP packets through the SIP server. When i am doing this, i have three PCs one is for Asterisk server. On another PCs two SJPhones are running. When i am calling from SJPhone to SJphone i can get all SIP messages through wireshark. But i am not getting not even a single RTP packets.
Can any one tell me what was the configuration mistake that i have done.
Regards,
Nagavardhan.
No. Because you haven’t provided copies of all the relevant configurations.
Now i got the solution.
When i mention the option directmedia=no then all the RTP packets passing through SIP server. I have a doubt that when this option i.e., directmedia=yes, no one RTP packets getting at server side.
I am using Asterisk server at the server side. even packets are moving between clients, the packets should appear at server side also.
Can any one tell me what is the correct procedure for that.
Regards,
Nagavardhan.
You are correct. If directmedia is set to no, all RTP packets run through the Asterisk server. If directmedia is set to yes, Asteriks tries to get out of the RTP path with SIP re-invites after the call is successfully established.
But only if there is no reason why it needs to see the speech or DTMF data.