SIP trunk

[cchaney]

Hey all -- thanks in advanced for even reading the post.


We just ordered a few SIP trunks from Bandwidth and have our vendor configuring the trunks with our 3300 PBX.

Up to this point, we are still having issues with one way audio -- they can hear us, we cannot hear them.  Test calls are usually placed from my teleworker phone and captures have been done using port mirroring and Wireshark.

Our PBX is NAT'ed behind a Cisco ASA security appliance, which is SIP aware.  From what I can see, it looks like our firewall is properly handling the translations/etc to/from Bandwidth's SIP proxy.

Bandwidth is reporting a ‘Not Acceptable’ from the PBX, from what they have said.



VoIP Flow:

|Time     | 10.1.2.2                              | 68.xxx.xxx.xxx                          |
|         |                   | 216.82.xxx.xxx    |                   
|0.000000000|         INVITE    |                   |                   |SIP From: "C.User" <sip:+281xxxxxxx@10.1.2.2 To:<sip:+1281xxxxxxx@216.82.xxx.xxx
|         |(5060)   ------------------>  (5060)   |                   |
|0.000449000|                   |         INVITE    |                   |SIP Request
|         |                   |(5060)   <------------------  (5060)   |
|0.097827000|                   |         100 Giving a try              |SIP Status
|         |                   |(5060)   ------------------>  (5060)   |
|0.097941000|         100 Giving a try              |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|4.205087000|                   |         183 Session Progress          |SIP Status
|         |                   |(5060)   ------------------>  (5060)   |
|4.205356000|         183 Session Progress          |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|19.623845000|                   |         200 OK SDP (g711U telephone-eventRTPType-101)          |SIP Status
|         |                   |(5060)   ------------------>  (5060)   |
|19.624424000|         200 OK SDP (g711U telephone-eventRTPType-101)          |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|19.672179000|         ACK SDP (g711U telephone-eventRTPType-101)          |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |                   |
|19.672840000|                   |         ACK SDP (g711U telephone-eventRTPType-101)          |SIP Request
|         |                   |(5060)   <------------------  (5060)   |
|26.235692000|         BYE       |                   |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |                   |
|26.236031000|                   |         BYE       |                   |SIP Request
|         |                   |(5060)   <------------------  (5060)   |
|26.346427000|                   |         200 OK    |                   |SIP Status
|         |                   |(5060)   ------------------>  (5060)   |
|26.346695000|         200 OK    |                   |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |




Here is the trace from Bandwidth support:

U 2012/11/06 20:32:48.834529 68.xxx.xxx.xxx:5060 -> 216.82.xxx.xxx:5060
SIP/2.0 406 Not Acceptable.
Via: SIP/2.0/UDP 216.82.xxx.xxx;branch=z9hG4bKb2b.d003737.0,SIP/2.0/UDP 67.231.xxx.xxx;branch=z9hG4bKb2b.5dbb3216.3,SIP/2.0/UDP 192.168.37.68:5060;branch=z9hG4bK0bB86c148cbb1cb7710.
Record-Route: <sip:216.82.xxx.xxx;lr;ftag=gK0b07b3aa>,<sip:67.231.xxx.xxx;lr=on;ftag=gK0b07b3aa>.
Warning: 399 10.1.2.2 "19H Reject".
From: <sip:+1919xxxxxxx@192.168.37.68;isup-oli=0>;tag=gK0b07b3aa.
To: <sip:+1281xxxxxxx@67.231.xxx.xxx>;tag=0_2240970064-98773126.
Call-ID: 1863043143_16435119@192.168.37.68.
CSeq: 17638 INVITE.
Contact: <sip:68.xxx.xxx.xxx>.
Server: Mitel-3300-ICP 11.0.1.26.
Content-Length: 0.


Happy Thanksgiving!

[martyn]

Removing the teleworkers as a problem, do calls work just from a local hand set on site?

[cchaney]

Going to have a user do a test call today and make sure a capture is taken during the same time -- I'm remote to that office so I'm reliant on my TW.

[martyn]

I wouldn't be surprised if it is just the teleworkers from what you have described. It would probably be a routing issue on the MBA server. Some static routes would fix it if that is the case.

[cchaney]

Ok so verified -- local users are able to call out using the SIP trunk but TW phones still not able to (remote party can hear, calling party cannot hear) -- and each time I can, I get the 'major alarms' from the MBG showing one way audio.

Could you give me an example of the static route configuration for the MGA/MBG that would allow this to work?

So happy you have me on the right path!

[cchaney]

And FYI, the MBG is setup using the Gateway mode profile

[x-man]

I wonder if its your router (at teh TW end causing the problem. SIP awareness can be a real problem area. Try turning it off locally and see if that makes any difference.

[martyn]

And FYI, the MBG is setup using the Gateway mode profile

Yep, guessed this to be the case. You need to add the address range for the sip provider to your local LAN list on the mbg , and tell it to go via the ASA. It will probably be trying to go via the MBG wan interface at the moment.

I'm on my iPad at the moment so can't look up the details, but I do remember seeing an interop doc on MOL the other day that has the ip range info in it.

[martyn]

Sorry I just noticed my earlier posts said MBA rather than MBG, as the iPad was doing auto correct.

[cchaney]

Thanks for the details.. will play around with it and see since don't have access to MOL.

[cchaney]

Tried a couple ways without success.  If you have a chance to review the interop MOL and post the items, that would be greatly appreciated.

Would the MAS have to be rebooted to apply these changes or should they work after applying?

And didn't mention previously, but this is a physical MAS and not virtual.

[martyn]

Yep no worries, I will have a look in the morning when I get in to the office, it's Sunday night here now, so can't be bothered getting the laptop out, enjoying the cricket and a beer too much!

You can't have a virtual mas in gw mode anyway, so all good.

[martyn]

ok, can you do a trace route from the shell of the MBG server to 216.82.224.202 and see if it goes via the public interface of the MBG, or via the ASA?

As a side note, I noticed that the call that you had in your original post was running uncompressed. It might pay to have it compressed unless there is a large amount of bandwidth at each end of the connection.

[cchaney]

Whenever I SSH into the MAS, I get the 'server console' -- how can I drop down to a regular bash to be able to traceroute/etc?

[martyn]

login as root rather than admin. Same password.