I have been using siproxd for years without problems. Without any changes to my system, calls from either of the two Minitar ATAs to one region result in no audio at the calling end but the called party can hear the caller. All other regions work perfectly.
Whilst I was debugging this I managed to establish one working call to a friend in the area in question. I noted that the failing calls all used another media gateway than the one I got through on.
A tcpdump on the external NIC on my firewall shows that all UDP packets sent from the "bad" RTP host result in ICMP port unreachable messages being returned from my firewall to that host. The ports shown as unreachable are those that I can see offered in the SIP session.
Is this too weird or is there a clue I missed?
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What version of siproxd are you running?
Is the behavior the same with the current snapshot of siproxd?
Could you provide such a tcpdump capture plus debug output of siproxd for such a failing call? You may send them to my SF email account.
If you would like to refer to this comment somewhere else in this project, copy and paste the following link:
Anonymous
-
2010-05-18
You are going to be saved some debug time because I was told to get the system working again LIKE NOW.
So, luckily we had some unused (at the moment) public IPs so for now there is no NAT in the equation and the users are happy.
The previous setup was running siproxd 0.5.13 on OpenBSD and had worked for a couple of years without issues except that the VSP early last year had a crazy media gateway on an AT&T IP (geoIP says in NJ USA!) and we are in Australia. It did the same thing as far as one-way voice is concerned but the VSP dropped that host about 15 months ago.
Anyway I noted a new version of libosip and siproxd (0.8) in the ports list yesterday and I'll be compiling an OS beyond the just released 4.7 (officially on 19 May, but I have mine already). I need to be at current to get to test the new SIP ports and to run the code I have had committed for bgp stuff.
So when all that gets together I'll build a new firewall with new sip stuff and try it on a weekend when nobody is looking.
Thanks for prompt attention and a product that has served us well for a long time.
If you would like to refer to this comment somewhere else in this project, copy and paste the following link:
I have been using siproxd for years without problems. Without any changes to my system, calls from either of the two Minitar ATAs to one region result in no audio at the calling end but the called party can hear the caller. All other regions work perfectly.
Whilst I was debugging this I managed to establish one working call to a friend in the area in question. I noted that the failing calls all used another media gateway than the one I got through on.
A tcpdump on the external NIC on my firewall shows that all UDP packets sent from the "bad" RTP host result in ICMP port unreachable messages being returned from my firewall to that host. The ports shown as unreachable are those that I can see offered in the SIP session.
Is this too weird or is there a clue I missed?
What version of siproxd are you running?
Is the behavior the same with the current snapshot of siproxd?
Could you provide such a tcpdump capture plus debug output of siproxd for such a failing call? You may send them to my SF email account.
You are going to be saved some debug time because I was told to get the system working again LIKE NOW.
So, luckily we had some unused (at the moment) public IPs so for now there is no NAT in the equation and the users are happy.
The previous setup was running siproxd 0.5.13 on OpenBSD and had worked for a couple of years without issues except that the VSP early last year had a crazy media gateway on an AT&T IP (geoIP says in NJ USA!) and we are in Australia. It did the same thing as far as one-way voice is concerned but the VSP dropped that host about 15 months ago.
Anyway I noted a new version of libosip and siproxd (0.8) in the ports list yesterday and I'll be compiling an OS beyond the just released 4.7 (officially on 19 May, but I have mine already). I need to be at current to get to test the new SIP ports and to run the code I have had committed for bgp stuff.
So when all that gets together I'll build a new firewall with new sip stuff and try it on a weekend when nobody is looking.
Thanks for prompt attention and a product that has served us well for a long time.