05-24-2021 03:52 AM
Greetings!
I am having problems with one way RTP streams on inbound calls.
My setup is:
ITSP->CUBE<- SIP TRUNK ->CUCM->UCCX
This setup was working before and all of a sudden when I try calling from my mobile phone, the signaling works, the call gets connected but I get no audio whatsoever meanwhile the other party can hear me.
This only happens on calls being generated from outside to the inside IP Phone, meanwhile for calls generated outbound everything works just fine.
I would greatly appreciate your responses and help.
EM
05-24-2021 04:03 AM - edited 05-24-2021 04:41 AM
Since you are not facing an audio issue for calls made from inside to PSTN, Routing issue is of less chance.
Is there a firewall in the call flow ? Can you explain a bit more about the setup. if firewall comes in to picture you need to check firewall configurations.
05-24-2021 04:12 AM
One way audio most of the time is a routing problem. So make sure that you don't have routing flapping causing intermittent RTP failure.
Another common reason (which can be your case) is a firewall device inspecting your RTP/SCCP/SIP traffic. I suggest to make sure that you look at inspection configuration and make sure that VoIP protocols are excluded.
Pls rate if its “Helpful”. If this answered your question pls click “Accept As Solution”.
05-24-2021 05:03 AM
Hi Sadav!
I can confirm that there is no firewall in between inspecting our RTP traffic.
It would be kind of surprising for me if it is a routing problem since calls initiated by our side are working fine, meanwhile those initiated from outside won't work as they are supposed to, I think @Nithin Eluvathingal is right about that part.
Anything else that comes to your mind about troubleshooting this issue or that you have faced before?
Thank you a lot for your time and patience!
05-24-2021 05:39 AM
Can you share the gateway configuration.
05-24-2021 06:01 AM - edited 05-24-2021 06:21 AM
Hi Ermir,
I agreed with Nithin if outbound working fine then there is less issue on routing part.
As suggested by Nithin pls check with your firewall team may be there is some rule configured.
I have faced same issue where my firewall team added some rule on ASA like they blocked the RTP traffic which going to outside and allowed Traffic for inside after raising a case with TAC they took on PCAP and suggest the same.
As per your setup call coming on UCCX environment am I right ? so cloud you please confirm how agent taking calls like via CTIOS or Finesse or any other application ?
Is agent taking calls as a normal agent or mobile agent if as a mobile agent then Nailed connection or call by call ?
Pls rate if its “Helpful”. If this answered your questions pls click “Accept as Solution”.
05-24-2021 06:45 AM
@Nithin Eluvathingal Yes I will attach it right now.
@Sadav Ansari I am currently sending the calls directly to the Operator's DN so I'm not using UCCX anymore, so that can't be on the way right now.
For any additional file or info or explanation please feel free to ask me.
05-24-2021 06:54 AM - edited 05-24-2021 05:26 PM
Can you remove the bind commands from incoming dial-peer 1 and other incoming dial-peer configured and try. I hope that could resolve your problem.
Since you bind the media/control on gi0/0. ping endpoint ip from gateway with source interface gi0/0. Most probably you cannot ping. and that could be the cause of one way audio.
05-24-2021 09:21 AM
Your Dial peer 1 communication bind on port 0/0
so could you please check is there any error on port if yes then pls clear counters .
Also there is ACL applied on your interface
“Ip access-group WACTH_TRAFFIC_FROM_ITSP_in”
Please check it once and also check the thing which suggested by Nithin.
interface GigabitEthernet0/0
description ==LINK to AI ITSP TRUNK ==
ip address 172.28.x.x 255.255.255.248
ip access-group WATCH_TRAFFIC_FROM_ITSP in
duplex auto
speed auto
05-25-2021 01:11 AM
@Sadav Ansari @Nithin Eluvathingal I tried taking off the bind interface from the interface connected to the ITSP, I also deleted the access list on that interface but the call now didnt get connected after i took the bind media command from that interface, It didn't help the case.
I tried pinging from the source interface as Nithin said and the ping worked.
05-25-2021 01:50 AM
What does it mean by bind command from interface?
It has to be removed from dial-peer 1
05-25-2021 01:54 AM
Sorry if I was unclear, this is what I meant to say:
dial-peer voice 1 voip
description === INBOUND CALLS FROM A1 ISTP===
no voice-class sip bind control source-interface GigabitEthernet0/0
no voice-class sip bind media source-interface GigabitEthernet0/0
05-25-2021 02:25 AM
05-25-2021 01:38 PM
Yes I have tried this, but it after applying this config the calls don't get connected anymore because the media and control aren't bound to any interface.
05-25-2021 09:14 PM
Reading your initial post again one doubt, the one way audio issue is happening just with your mobile phone or for all incoming calls. i used to configure bind commands for outbound dial-peers but i don't remember anything i configured on inbound directions.
Check if you are able to reach the internal end point ip from your bind interfaces. if that works you need to do packet capture and see if you are getting the RTP packets.
second thing you mentioned the problem started suddenly, what changes happened in the network ?
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