CME to CME over internet behind NAT/PAT on both sides scenario

davenull

JeanM wrote: »

My next plan of action is to extend this lab over the internet, by finding another person who is willing to configure a CME router and dial-peer. That way it will really be over internet WAN .

Let me know if you are interested to give it a shot.

I would like to give it a shot. I'm still at the beginning of my voice studies, but I should be able to mirror your config.

JeanM

davenull - cool! PM me when your side is configured and we'll exchange IP addresses and dial-peer info.

Jollycork

So were you two able to get the CME's to talk to one another?

JeanM

Haha Yeah I am waiting for davenull to PM me once his side is ready so we can test this out. Or if someone else wants to give it a shot, just configure your CME the same way and PM to exchange the internet facing ip and dial-peer info.

davenull

I just set up an 1841 as my edge router and re-purposed my SOHO router to act as a WAP - can't leave folks at home without internet

I'll set up the 2811 with CME tomorrow after work.

Sorry to keep you guys in suspense. The work has been busy - I was sidetracked into having to study powershell.

JeanM

Sweet. I did something similar with 1841 + tplink ap .

I'll ping you tomorrow after work.

JeanM

Hey, I must have missed you when you were up. Ping me your email address, as this forum didn't notify me when you send me an IM.

JeanM

So Update.

Myself and davenull worked on this, and got voip to voip calls to work between two CME routers each one behind edge/nat router over the internet.

So, interestingly, we got h323 calls to work but could not get sipv2 to work. Another thing is we could not get g729 codec to work either, but g711ulaw and alaw and also g728 worked for both of us.

In my lab, I got sipv2 working just fine using the same cme-nat-wan-nat-cme topology.

Anyways, so we got the calls to work (including caller-id) but would like to know why g729 didn't work with h323 or sip.

Now that I think about this, maybe sipv2 would work if we tried g711/g728 on it like we did with h323!


Also, did not expect that #debug ccsip calls would show h323 calls ? Example below.
Does the h323 fall under SIP realm , since it was popular before SIP etc?




Jan 27 04:01:50.262: //35/ECD9769F8028/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x83A9B56C
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 2001
Called Number : 7701
Source IP Address (Sig ): 192.168.1.99
Destn SIP Req Addr:Port : 24.240.xx.xx:1079
Destn SIP Resp Addr:Port : 24.240.xx.xx:55375
Destination Name : 24.240.xx.xx


Jan 27 04:01:50.266: //35/ECD9769F8028/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0
Source IP Address (Media): 192.168.1.99
Source IP Port (Media): 17090
Destn IP Address (Media): 24.240.xx.xx
Destn IP Port (Media): 16620
Orig Destn IP Address:Port (Media): 0.0.0.0:0


Jan 27 04:01:50.266: //35/ECD9769F8028/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 200

Jollycork

not sure why that particular codec wouldn't work,

Jollycork

Just curious .

JeanM

flowroute, per another forum member recommendation

davenull

Jean, did I need to sign up for a sip provider too? I never signed up for any.

JeanM

No, that's for a different lab. That way you can connect your router with external sip provider like flowroute and make calls outside to PSTN/cellular or if you purchase DID(s) can make calls into your lab.

Jollycork

hum maybe something to try to work on translation no's and DIDs in CUCM...

I just hook up my CUCM to POTS and use a connection PLANAR to Unity...

might be fun to use SIP with DIDs

If I get SIP setup into CUCM with DIDs [maybe caller IDs] I'll let you both know, we can test calling over SIP into CUCM on DIDs [provided the wife oks leaving the lab on (electric bill goes up, up, and away if I leave it on).

I've got 3 remote locations setup over frame to my HQ CUCM, be fun to see if you guys and call into the remote sites over SIP