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Troubleshooting SIP Audio Problems

Started by RonR, February 18, 2012, 12:32:57 PM

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RonR

Aaron Clauson, the creator of SIP Sorcery has posted a SIP and Audio Guide 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.

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.

carl