SIP calls are dropping - 06/15/11 11:02 PM
Okay all you sip experts...help!!
So we just set up a new MBX with IP phones. Customer elected to go with SIP as a money saving option due to its a small office. Everything appears to work normally internall, but 1 out of every 10-15 calls is getting dropped. Its random.
The SIP provider is a small local company and we have been having issues with support. I did a packet capture using wireshark and sent the results to Vertical. The only real discrepancy they could find was that the provider seemed to be trying to change the RTP port number in mid call. This didn't seem right to them.
So my understanding when it comes to SIP, in a nutshell, is that the connection for the call is established using UDP 5060, and that the actual voice audio is transmitted using RTP over one of the higher ports in the router? Did I get this right? Its not too far off of the same principle of a PRI and B-channels and D-Channel. They seemed to have a large number of ports opened up for RTP.
So anyway my questions are..
If the call is established and the audio is given port 50555 to transmit on, that port should never change, right? Why would the provider be sending a request to our PBX to change the port in mid-call?
Is the SIP connection on 5060 only from the MBX system to providers router?
Does the packet capture only concern itself with data flowing from the MBX to the SIP providers router, or does it capture all the data that is being sent from point A, the caller, to point B, the person being called?
Thanks in advance for the help.
So we just set up a new MBX with IP phones. Customer elected to go with SIP as a money saving option due to its a small office. Everything appears to work normally internall, but 1 out of every 10-15 calls is getting dropped. Its random.
The SIP provider is a small local company and we have been having issues with support. I did a packet capture using wireshark and sent the results to Vertical. The only real discrepancy they could find was that the provider seemed to be trying to change the RTP port number in mid call. This didn't seem right to them.
So my understanding when it comes to SIP, in a nutshell, is that the connection for the call is established using UDP 5060, and that the actual voice audio is transmitted using RTP over one of the higher ports in the router? Did I get this right? Its not too far off of the same principle of a PRI and B-channels and D-Channel. They seemed to have a large number of ports opened up for RTP.
So anyway my questions are..
If the call is established and the audio is given port 50555 to transmit on, that port should never change, right? Why would the provider be sending a request to our PBX to change the port in mid-call?
Is the SIP connection on 5060 only from the MBX system to providers router?
Does the packet capture only concern itself with data flowing from the MBX to the SIP providers router, or does it capture all the data that is being sent from point A, the caller, to point B, the person being called?
Thanks in advance for the help.