Wondering if anyone can help me.

Our head office is running an Avaya phone system and all mikrotik routers. We have a couple of remote locations with VPN tunnels between us. We gave them some IP Phones to use. When we call them or they call us I can only hear them speak and they can't hear us. Ping times between us are not too bad -- average around 20ms. When people at the same office call either other (located on the same LAN) it works fine. Avaya is telling me its a network issue and they won't help me. They told me to look into QOS but my logic is -- if it was a packet prioritization problem wouldn't I be able to hear them a little? I understand it might cut in and out but I should hear something. Suggestions??

In addition to my original post....

Found this in the Avaya manual

End-to-End Matching Standards:
VoIP depends upon the support and selection of the same voice compression, header compression and QoS standards throughout all stages of the calls routing. The start and end points must be using the same compression methods. All intermediate points must support DiffServ QoS.


How do I know if all my intermediate points are using/support DiffServ QOS?

You might want to look into the tehnical specs of hte Avaya AP-8 (it's a rebranded Proxim AP4000 with upgraded firmware, and some QoS) to see if they mention any specifics about the QoS on the AP feature/spec for comparison. I recall that proxy-arp has to be enabled on them - might want to try that on your MT...

Figured it out!!

I had a src-nat rule to our remote office from our network - once I disabled that the phone works both ways.

i have exactly the same problem ,,, i could not solve it ... would you please tell me how could you solve it exactly
Best Regards
Mahmoud Kamel

i have exactly the same problem ,,, i could not solve it ... would you please tell me how could you solve it exactly
Best Regards
Mahmoud Kamel

Basically one way voice is usually due to a firewall rule blocking packets, a NAT rule changing packet addresses to something the endpoints don't expect, or maybe you have the SIP ALG active and it's interfering.

ip > firewall > service ports : make sure sip is disabled.

if your phones are to communicate through a VPN, make sure that there is no NAT in either direction from branch to HQ, or from HQ to branch.

make sure you are not blocking traffic in a way that would discard the RTP sessions. (If phone A hears phone B, but B cannot hear A, look for filters that drop packets from A towards B.)

If the PBX support tells you this is a QoS issue, they just want to get off the phone and close the ticket. QoS issues make the audio quality garbled and bad, but NOT dead silence. Dead silence comes from bad signalling (inspect SIP messages to see what IPs the phones are instructed to send/receive audio from), packet filtering, or improper settings having to do with NAT translation.

Thank you very much for you reply ,,
In fact i have very complicated network
i have internet leased line with physical VPN line that connect 2 sites
the internet line getaway is 192.168.0.5 . the VPN router getaway is 192.168.0.1 they are in the same LAN that connected physically to other location.. The other location getaways is 192.168.1.1 and 192.168.2.1 ..
The avaya is connected through these vpn .. i tried all possible solution on the internet but unfortunately, didn't succeeded still the voice go in one direction i heard them but they don't hear me

Thank you very much for you reply ,,
In fact i have very complicated network
i have internet leased line with physical VPN line that connect 2 sites
the internet line getaway is 192.168.0.5 . the VPN router getaway is 192.168.0.1 they are in the same LAN that connected physically to other location.. The other location getaways is 192.168.1.1 and 192.168.2.1 ..
The avaya is connected through these vpn .. i tried all possible solution on the internet but unfortunately, didn't succeeded still the voice go in one direction i heard them but they don't hear me

I recommend that you eliminate possible causes - make sure that the phones are being given the right media address when connecting calls. (packet capture a call from a point that is as close as possible to the phone that can hear the other phone) Look for the SIP connect messages and read the SDP portion to see what address / port the media should be sent to. Make sure the phone isn't being told to send the audio to the wrong place.

When you know both phones have correct IP where to send the media (your config might use the PBX in the middle for all calls, instead of direct phone-to-phone, so don't be surprised if you see this)

Once you know where the media is intended to go (if correct) then make sure the routing is correct, and then try capturing packets at various points along the path to see where the drop happens.

Then when you know WHERE it drops, examine things like routing tables, firewall rules, policy routing, NAT, Sip ALG/protocol helper, etc to find why.

Good luck!