I didn't receive any response, but I'm still struggling with this issue
- The problem is intermittent, making it difficult to troubleshoot. At
least I've now managed to capture a tcpdump of a problem call.
I am seeing ICMP port unreachable messages from the server to the phone
when the one way audio occurs.
This is my current theory, but I'm not sure it makes sense.
Somewhere during a communication with the server, either the server or
phone misses a SIP message and so one of them still believes it is on a
call.
Next outbound call made by the phone, when the phone sends the RTP
stream, it is being sent to the old UDP port on the server from the
previous conversation. The server rejects this with an ICMP port
unavailable as it is expecting RTP from the phone on another port. This
results in one way audio - the external person being called cannot hear
the callee.
When an inbound call is made, the server initiates the RTP ports with
the phone and two way conversation is heard.
Does this make any sense? I have a dump file of the conversation if
anyone is interested?
Is anyone else seeing UDP issues with Asterisk 1.2.23 or even 1.2.21?
Any thoughts on trying to determine what is causing this?
Thanks
Martin Glazer wrote:
Hi All,
I'm experiencing a problem with intermittent one way audio on some
outbound calls. We are not using NAT at all and have an incoming T1
line for external connectivity. Asterisk is running on a dedicated
ethernet network without other traffic.
I'm using about 35 Aastra SIP phones for the clients and Asterisk
1.2.23 on the server.
Any suggestions on trying to track this down? I'm actually suspecting
network connectivity issues but don't know how to proceed.
Any help appreciated.
Thanks
Martin
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