CUCM one way audio communication help request.

[mabuzaid1]

Dear Community members ,

Hope you all are doing great in your life and professional jobs .

I am having an issue with the Call manager version 10.5 and 3rd party GSM Gateway (Sip trunk).

We have an extension 6876 that suppose to call to outside network (national / international ) number when pressing *7.

Also the outside network can call to my network (GSM Sim number ) which should direct the call to my voip extension at CUCM .

Now we tested the call from the Voip extension at CUCM to call PSTN numbers its working .

Also We can call from PSTN to GSM gateway (Cell number at GSM Gateway ) it directs the call to Voip extension just fine .

Our problem is when ever the call is established between PSTN / CUCM extension  we can hear only one way audio .

The participant on CUCM can hear the voice of PSTN guy but the PSTN number never hear the guy on CUCM audio .

no matter who calls whom .

---

Notes:

• The SIP Trunk (GSM Gateway is from 2N vendor ) , CUCM Version 10.5 .
• The Voip phone is on the same partition and calling search space as for the GSM GW.
• No media resource is assigned to the phone / extension .
• does the device pool has any thing to do with this ?
• if this is due to media resource doesn't the call suppose to go to the call GSM GW in VOIP (it shouldn't do     trans-coding or changing the format to PSTN Q.931 format as i understand ?.
• The client has other Cisco Voice GW which should not involve in this route pattern .

Thanks and willing to hear from you guys .

Regards

Mansour

[Slavik Bialik]

One way audio is happening mostly because of:

1. You're lacking of a route inside your network. Meaning, that the phones cannot send their RTP to the 3rd party SIP gateway which you are connected to it with a SIP trunk. Verify that you have that route. 
   Note: I would verify which route you must configure with reading the SDP in the SIP messages, just check which IP address is returned from this PSTN gateway. Most of the times the PSTN have media gateways that are handling the RTP, and SIP gateways that handling the signaling. So maybe your issue is that you only have routes to the SIP gateway and not to the media gateways handling the RTP.
   Use RTMT tool in order to view the signaling and the SDP negotiation.
2. Firewall policy - maybe you have a firewall in your organization that is blocking the RTP packets from the phones (or from CUCM, if you enabled MTP on the SIP trunk that is going towards PSTN) to your PSTN provider relevant servers.

I can't see any other logical reason why you'll have one way audio.

[mabuzaid1]

Dear Slavik ,
Thanks for your reply .

Well , in the 3rd party gateway we have 2 ip address as following :
- The Gateway CPU IP address (responsible to establish the call as i understood from the datasheet (SIP signaling )
- The Voip Card on the same unit (resposinble for RTP packets , which as i understood from a very limited documentation ,
It routes the calls
Ok , if we assume that SIP is established and we have only one way audio (CUCM can't rout to the VOIP card to handle the RTP ,
Where in Cucm i can configure the RTP Connection (Voip Card ) IP Address .

I mean can i tell the CUCM to use for example IP Address :1.1.1.1 /24 for SIP negotiation .and for RTP to go to 1.1.1.2 /24 ?
Is it possible ?
note:
In GSM SIP Trunk we have configured the CUCM IP 1.1.1.3 /24 to register and Voip card 1.1.1.2 to handle the RTP as per their manuals
And i alaso engaged a technical remote session form the vendors and all seems to be OK from GSM side .

I will consider also the firewall configuration with desired technical person .
Thanks in Advance .

[Slavik Bialik]

Hi,

Is CUCM and this GSM Gateway are located on the same LAN? If that's the case, it is not firewall and not network routes.

If not, keep checking this thing.

By the way, at first I thought this GSM gateway is already located on PSTN side, and not yours. 

And not, you cannot tell CUCM what is the IP address of the RTP card of your GSM gateway, as it is not necessary. Because your GSM gateway should present this IP address by itself when making the SDP negotiation. So please check also the SDP messages and see if your gateway presenting the correct IP address (1.1.1.2) in the SDP.

Besides that, the issue can be on the GSM gateway itself that can result of miss-configurations. 

So I would also put some Wireshark capture on a test phone (using spanning to PC port) and see if the phone is sending back RTP packets, and if so... where. If it does send the RTP correctly to the relevant IP, I would check the GSM gateway somehow (like captures if possible) as it is the gateway that's connected to the PSTN as much as I understand. Maybe the issue is that the GSM gateway cannot send the RTP from the VoIP card to the IP addresses of the PSTN.

[mabuzaid1]

Thanks again Slavik for your valuable input .
I guess it will be something related to VOIP card routing the RTP to / from CUCM .
Below link is from the vendor side of GSM Gateway (which is in my same LAN "both CPU & VOIP " and ping-able ).

https://wiki.2n.cz/btwsgum/latest/en/2-description-and-installation/2-1-plug-in-boards/voip-board

But as i understand from you that either CUCM is not forwarding the RTP TO GSM Voip CARD or the VOIP Card is not forwarding the RTP To PSTN .
So shall i use the RTMT (at my pc connected to cucm extension and see where the cucm is forwarding the packet ?)

Thanks

[Slavik Bialik]

If you didn't enable MTP on the SIP trunk towards the GSM gateway, the RTP flow will come directly from the phones towards the GSM gateway. If you did enable MTP, the RTP flow will be: Cisco Phone -> CUCM -> GSM Gateway -> PSTN.

So according the above statement, connect a PC to a Cisco phone to its PC port, and enable the "Span to PC Port". Open Wireshark on this PC, and make the following filter: sip || rtp (rtp is the most important of course). Then you'll see the packets incoming from the GSM gateway, and you must see the opposite packet flow, so just check if those packets are being transmitted to the correct destination IP address.

[mabuzaid1]

Will check that after tomorrow and will update you .

Thanks in advance .

[mabuzaid1]

I wanted to clarify this , That GSM gateway is having a sim cards (for PSTN side ) and the other side is VOIP through the switch to the CUCM side .
As i see in the GSM document , that it supports g.711 ,g.729 etc so i don't think that i need transcoding or MTP ?
As the GSM GW will handle the voip and convert it to from PSTN format .

Please advice on the MTP .
one more thing its not required to present cisco Voice gateway for the communication , Right ?(it will not be any added value ?)

Regards
Mansour