Yate Community Forum
Yate server => Other Yate server issues => Topic started by: yyzhang on March 07, 2014, 02:41:18 AM
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I want to realize that the voice data(RTP package) can transport directly from client to client and do not pass through the yate server.
How can I configure the parameters in configuration files to realize this?
I hope there is somebody can help me solve this problem !Thank you very much!!!
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See http://docs.yate.ro/wiki/RTP_Forwarding
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Thank you for your reply,but that document is about h.323 client to sip client.Our system has just two sip client and do not involve h.323.
In this situation,how can i realize it?
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In the same way.
The document IS NOT about h.323 to sip interworking, even if the included image shows an example of h.323 to sip call.
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OK,I‘ll try it.Thank you very much!
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I read the document,is it mean that I have to put the line below in the regexroute.conf ?
[default]
${rtp_forward}possible=;rtp_forward=yes
but it doesn't work!
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If there is no rtp_forward=possible in call.route it means the caller didn't offered media.
Can you dump the call.route message?
And the received INVITE?
Better, can you dump a log with message sniffer and sip messages?
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I use wireshark and see the SIP signal and RTP package all pass through the yate server
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I can't help you without yate log.
RTP forward is working. If you say is not working I need to see the log to see what happens.
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Thank you for your help!I install yate server in ubuntu12.04 instead of windows,and modify the configure files as you post that link.It is successful!
Maybe the configuration is not effective in windows.
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You might not edit the right file on Windows.
See http://docs.yate.ro/wiki/Debugging_Yate_Client_on_Windows
See Introduction section.