No RTP Stream

[steveycarr]

Hi Guys,

I've got 3 Mitel 3300's all linked together with Sip Trunks in a Training Lab. Everything works great, I can call any extension from any switch etc, even in and out of the PSTN which I have on one of the switches.

I'm attempting to Wireshark Trace all the Sip and RTP traffic between the switches as part of a Training exercise. I'm mirroring the Mitel LAN side ports on a Cisco 3560 POE switch and I can Capture Sip traffic on all the ports but on two of the Mitel's I can't see the RTP stream. All the Mitel's have Voice Encryption turned off. If I capture the call on the Phones port on any 3300 I can see the RTP but not when I'm capturing on the Mitel LAN port.

The Mitel which is ok is Rls 4.2 and the other two troublesome switches are 7.1 and 8.0.

It's not the end of the world as its only to deliver some Wireshark training but I'd like to resolve it. It's got to be something simple that I've missed. Is there some other setting that needs invoking. Any ideas would be much appreciated.

Regards,

Steve

!ExtraDunce

 

[sarond]

What you're seeing is by design.

The Mitel is involved in the signalling only, RTP is direct peer to peer (phone to phone)

This is why it is important to have correct firewall rules and routing otherwise you can get one-way/no speech etc...

One of the purposes of having an MBG is to get around this by anchoring the RTP to it.

 

[steveycarr]

Hi Sarond,

Thanks for your reply, however I have to respectfully disagree, to start with there is no Firewall, its in Lab situation, secondly how come I can see the RTP stream on the Lan side of the Mitel with 4.2 sw. I don't have any one way Audio or speech problems everything works ok. The 3 Mitel's are connected to a single Router with 3 interfaces on 3 Subnets so the Subnets are directly connected so I've no Routing as such.

Listening/mirroring the LAN port of my 4.2 Pabx I can see both incoming and outgoing RTP streams from both of my other Pabx's, however listening LAN side on the other 2 Pabxs I cant see the RTP streams whether its to the 4.2 switch or between the other Mitel's. The Pabx's are the endpoints (peers) so I should be able to capture the RTP Stream between the 2 of them.

We could have a situation where a Mitel may just have analogue phones and have SIP Trunking without a MBG, its not mandatory to use a MBG. So in that case we must be able to analyse the RTP stream on the SIP Trunking. I've done this on a number of occasions and the only time I cant see/hear/analyse the RTP is when the Voice Encryption is enabled.

Regards,

Steve

!ExtraDunce

 

[Billz66]

Sarond is correct RTP is always peer - peer
4.2 is pretty old as far as software is concerned but from memory its always been peer to peer for RTP

the only time RTP traverses the MIVB is in the follwoing cases
- call to Onboard TDM device( ISdn , analog trunk Analog extension)
- MOH will stream from controller
- IP telephones connected to an onbaord Switch(Cxi)
- if 2nd controller is connected via 2nd lan interface of the MiVB (Mxe)

we would probably need to see a proper network diagram showing Ip address,subnet and gateway addresses of all devices( MIVb & phones) to understand whats going on

If I never did anything I'd never done before , I'd never do anything.....

 

[sarond]

Hi Steve,

I agree that you have no problems with firewalls, routing etc...

I also agree it's not always mandatory to have an MBG, it can just resolve a lot of issues. (Normally with remote devices)

Maybe a physical topology on what is connected to what ports and which devices, vlans etc... would assist.
Also what scenario are you testing. Internal calls, SIP trunk calls. Which controller are the SIP trunks connected to.

I'll admit I'm not an expert but I did't think RTP traversed the MCD unless required. i.e. TDM devices etc...
What does Wireshark show in the SDP, is it using the MCD address or IP Phone address?

 

[steveycarr]

Hi,

Thanks once again for your time, I've uploaded a diagram. So in a nutshell Mitel B is acting like a Sip Service Provider. Each Mitel having a Sip Peer profile pointing to the other.

Foe Example a call from X100 out to the PSTN works ok, no issues at all, same goes for inward DDI Call to X100. At Point B I can capture the Sip signalling and the RTP Stream and listen to the trace afterwards, however at point A I can only capture the Sip Signalling and I don't see the RTP stream being captured. The call is fine so the RTP is present I'm just not seeing it.

I'm sure I've captured RTP streams in the past between Mitel's and SIP SErvice Providers. How else are you supposed to troubleshoot audio problems?

Regards,

Steve



!ExtraDunce

 

[sarond]

Ok I can understand where it is getting RTP at point B as it have to go via the gateway controller.

Does point A have more to it, like a switch involved.

Are you mirroring the Mitel port in section A? If so I would try the mirroring the port connected to Gi0/0 on the router instead to see if you get RTP traffic as it has to traverse that port.

 

[sarond]

Maybe with previous port mirroring to SIP providers you captured the uplink port??

 

[steveycarr]

Hi Sarond,

I'm not sure what you mean by a Gateway Controller, there's just 2 Mitel 3300's connected through a Router. The LAN Port (uplink if you like) of each Mitel is connected via 3560 switches to the G0/1 and G0/0 ports on the Router. The 2 switches are flat and have no config and are only in the scenario so I can port mirror. The mirrored ports are the Lan/Uplink ports on both the Mitel Cxi's

So mirroring at point A doesn't capture RTP but the other side of the Router at point B it does capture the RTP, its the same stream on the same call at both ends. I just can't see/hear the RTP at both ends. I am capturing at both points A & B on the Mitel Lan/Uplink Ports before they go into the Router.

I always capture the uplink when troubleshooting sip/rtp when a Sip Provider is involved. I work on Avaya, Siemens ISDX, Hipath CS1K as well as Mitel and I've not come across a scenario when I cant capture the RTP unless of course its encrypted either by the Voice switch or some other device.

Regards,

Steve




!ExtraDunce

 

[sarond]

By Gateway I meant the controller connected to the eternal SIP trunks (B).

The port the Mitel is connected to is not necessarily the uplink port. The port connected to the router Gi0/0 is the uplink.

The reason I say this is because the phone RTP doesn't have to go via the Mitel but it does have to go via the router.
Mirror the port connected to the router Gi0/0 and see if you get RTP.

I understand when troubleshooting with a provider you would capture the uplink and all should be good. The Mitel is not the uplink though in this scenario.

 

[steveycarr]

Sarond,

Solved, all my own work I'm afraid. It doesn't help to be monitoring the wrong port. Found out by using 2 analogue phones, one at either end.

Many thanks for your help your first thoughts solved the issue in the end.

Regards

Steve

!ExtraDunce