Aaron Clauson, the creator of SIP Sorcery (http://www.sipsorcery.com) has posted a SIP and Audio Guide (http://www.sipsorcery.com/mainsite/Help/SIPAudio) that explains how RTP works with SIP and some of the common problems that can occur.
Aaron has also created a diagnostics tool that can be used to perform a cursory check of the ability to transmit an RTP packet to and from a single SIP device. To test an OBi's SIP/RTP operation:
1. Click the Go button on the SIPSorcey RTP diagnostics page (http://diags.sipsorcery.com/).
2. When presented with the Waiting for call to nnnnn@diags.sipsorcery.com prompt, dial the following from the OBi PHONE Port:
nnnnn*23*21*232*122#
Note: Prefix the above with **1 or **2, as necessary, if your PrimaryLine is not configured for SIP.
3. A response similar to the following should be displayed:
INVITE request received from udp:147.212.6.83:5060 for sip:nnnnn@23.21.232.122:5060.
Advertised RTP remote socket 147.212.6.83:16802, expecting from 147.212.6.83:16802.
RTP received from 147.212.6.83:16802.
Sending dummy packet to 147.212.6.83:16802.
Test completed. There were no RTP send or receive errors.
BYE request received from udp:147.212.6.83:5060 for sip:23.21.232.122:5060.
Thanks. A great info.