Sorry, I'm a complete VoIP novice...
We have a bit of a weird problem in that when you dial an extension which is our other office (connected by 2 x 2mpbs MPLSs) the caller can hear the recieving person's voice but the the recieving can't hear the caller. This happens both ways, i.e. when we call them and when they call us. We do have an access-list in place and I have opened up the ports that were logged as being denied when the calls were placed (namely 51000 range) and now nothing is being logged as denied.
Apparently this has worked when initially tested, so I'm not sure what could have changed.
Any ideas? Has anyone come across this before? We are using Avaya phones.
one way voip can have a couple of reasons.
1) filters - as described. VoIP ports are dynamically negotiated. For testing purposes remove the filters potentially blocking access between the phones.
2) IP routing - make sure there is connectivity between the phones. F.e. there could be a static route missing somewhere in one direction. Simple first check: can the default gateway of one phone ping the other phone. Again make sure filters are not interfering with your connectivity test.
3) QoS related - packets might be dropped because of an overload situation in one direction. This , however, is usually a transient issue, while the overload situation occurs.
4) Software bug - anything is possible. Can the phone in question be used with other locations?
Thanks for your response Martin, much appreciated.
1) I have removed all filters and re-tried: same result.
2) Definitely connectivity is OK, OSPF routes in place, lots of traffic passes between the 2 regularly. Verifyied by ping as suggested.
3)This could be an issue...I've looked at the interfaces in question, and some of them have a large amount of drops - input queues leaving from here and output queues from the other offices, e.g.
Output queue: 0/1000/64/1067272 (size/max total/threshold/drops)
However, I have cleared the counters and tried again, with same results, yet no more packets had been dropped.
4) We currently only have 4 IP phones (1 at this site, 3 at other) as we are still setting it up. The problem is with all 4 phones, when going between offices. They work fine within the same office (to IP and standard analogue phones). Also, I can use the IP phone here to dial out to my mobile with no problems (this does not traverse the 2mpbs MPLS).
We will be implementing some QoS queueing tools but I was under the impression that we should be able to get at least some, albeit choppy, audio working. Do you think it's likely/reasonable that this problem won't go away until proper queuing is implemented?
Many thanks for your help.
In the phone where they are not hearing, can you press 'i' button twice to get call statistics. Check Recd Packet, Recd Discard and Recd Lost. If you don't see Recd Packet (which I suspect), it either IP Routing or ACL blocking.
I have had that problem too with IP phones on 2 sides of a VPN connection. The way we had to get around it was to manually populate the ARP tables on the Communication Server side. You can check to see if you have entries that match your hardware with a SH ARP.
I have had it referred to as One Way Audio.
There is a white paper on it. I will try to find it an post it.
Thanks for the replies guys.
Chris - the gateway routers are both 3750s running 12.2(25r)SEC.
pcogdell - there is only an acl on one end, and I completely removed it and had the same results, so I'm hoping that I can count that out....
dmalloch - it's not a VPN connection but I guess the same thing can apply... I will definitely look into that.
Are the switchports configured as access or trunks? If the switch is not configured correctcly and you are marking the frames with 802.1p (COS) the switch may drop the frames sent from the phones.
Try this on the switch ports connected to the handsets.
switchport voice vlan dot1p
Well they are connected directly to the switch (no PC) so I had put them in the normal vlan. Anyway I've just tried the command and it didn't change anything unfortuntely.
I would have thought that if that was the case I wouldn't be able to use the phone at all, and yet I can dial within this site (nonIP) and externally with no problems....
Anyway thanks for trying, I appreciate it.
Could you send the configuration setting of your interfaces and sub-interfaces between the two gateways. when we have encountered one-way audio it was cuased by header-compression enabled on one side or fragmentation mismatch on the two links. (the later would cause data interrupts also.