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Users behind NAT


dslepnev
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Hi,

 

We trying to configure pbxnsip to work with remote users who would like to work at home. Most of remote users have internet connection behind NAT. It causes problem when users don't hear audio at all. What happens:

 

This is INVITE from remote user. This INVITE coming from real address.

1 0.000000 195.146.74.104 192.168.22.43 SIP/SDP Request: INVITE sip:112@tw;user=phone, with session description

pbxnsip answering to this address. Good.

2 0.003732 192.168.22.43 195.146.74.104 SIP Status: 100 Trying

3 0.010382 192.168.22.43 195.146.74.104 SIP/SDP Status: 200 Ok, with session description

and trying to send RTP to private address. This address taken by pbxnsip from INVITE.

4 0.021485 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24774, Time=3241200, Mark

6 0.041015 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24775, Time=3241360

 

Sure that pbxnsip working fully according SIP RFC - sends RTP to IP address from INVITE/SDP.

 

Quesion is: maybe possible to configure pbxnsip to send RTP to addres from INVITE received? As part of Mini-SBC function..

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Hi,

 

We trying to configure pbxnsip to work with remote users who would like to work at home. Most of remote users have internet connection behind NAT. It causes problem when users don't hear audio at all. What happens:

 

This is INVITE from remote user. This INVITE coming from real address.

1 0.000000 195.146.74.104 192.168.22.43 SIP/SDP Request: INVITE sip:112@tw;user=phone, with session description

pbxnsip answering to this address. Good.

2 0.003732 192.168.22.43 195.146.74.104 SIP Status: 100 Trying

3 0.010382 192.168.22.43 195.146.74.104 SIP/SDP Status: 200 Ok, with session description

and trying to send RTP to private address. This address taken by pbxnsip from INVITE.

4 0.021485 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24774, Time=3241200, Mark

6 0.041015 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24775, Time=3241360

 

Sure that pbxnsip working fully according SIP RFC - sends RTP to IP address from INVITE/SDP.

 

Quesion is: maybe possible to configure pbxnsip to send RTP to addres from INVITE received? As part of Mini-SBC function..

 

 

I have faced the same situation, to resolve I didi the following:

 

1- use a router that support DMZ and set to the adress of your PBX server - work fine

 

OR

 

2- set your router to allow ports for your PBX - all of them RTP is transporting audio

 

you can also set your PBX to use the following route table :

under Admin>Ports set "SIP IP Replacement List" to first IP of your PBX and IP of your router - something like 192.168.0.1/xxx.xxx.xxx.xxx -> PBXIP/RouterIP

 

restart PBX and It should work fine

 

This run fine on my side

 

BTW you should have a very very very good Internet connection if you want to keep audio quality for all external registration on your PBX otherwise it will be more like a cell phone quality, forcing codec might help.

 

hope it help!

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Hi,

 

We trying to configure pbxnsip to work with remote users who would like to work at home. Most of remote users have internet connection behind NAT. It causes problem when users don't hear audio at all. What happens:

 

This is INVITE from remote user. This INVITE coming from real address.

1 0.000000 195.146.74.104 192.168.22.43 SIP/SDP Request: INVITE sip:112@tw;user=phone, with session description

pbxnsip answering to this address. Good.

2 0.003732 192.168.22.43 195.146.74.104 SIP Status: 100 Trying

3 0.010382 192.168.22.43 195.146.74.104 SIP/SDP Status: 200 Ok, with session description

and trying to send RTP to private address. This address taken by pbxnsip from INVITE.

4 0.021485 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24774, Time=3241200, Mark

6 0.041015 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24775, Time=3241360

 

Sure that pbxnsip working fully according SIP RFC - sends RTP to IP address from INVITE/SDP.

 

Quesion is: maybe possible to configure pbxnsip to send RTP to addres from INVITE received? As part of Mini-SBC function..

 

Based on the above log, it looks like SIP messages are not reaching the phones. I do not see an ACK from the phone for the 200 OK. Please make sure that the home router is not blocking the traffic from PBX. Generally, SIP/RTP uses UDP. So you have to make sure that port UDP 5060 is not blocked and for RTP, 49152-64512 are not blocked. Please refer Admin->settings->ports page to see these values.

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Based on the above log, it looks like SIP messages are not reaching the phones. I do not see an ACK from the phone for the 200 OK. Please make sure that the home router is not blocking the traffic from PBX. Generally, SIP/RTP uses UDP. So you have to make sure that port UDP 5060 is not blocked and for RTP, 49152-64512 are not blocked. Please refer Admin->settings->ports page to see these values.

 

SIP messages reaching ip phones.

I checked that home router do not blocking the traffic from pbx. Pbxnsip send media traffic to internal IP address, and this media can't be delivered to the ip phone. This is the problem.

 

Sorry, I can't attach the trace "Upload failed. You are not permitted to upload this type of file".

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I have faced the same situation, to resolve I didi the following:

 

1- use a router that support DMZ and set to the adress of your PBX server - work fine

 

I need to set routers which supports DMZ on all user's locations. Not best way. SIP ALG?

 

OR

 

2- set your router to allow ports for your PBX - all of them RTP is transporting audio

 

There is problem not in ports. Problem in addresses, which coming inside SIP messages to the PBX.

 

you can also set your PBX to use the following route table :

under Admin>Ports set "SIP IP Replacement List" to first IP of your PBX and IP of your router - something like 192.168.0.1/xxx.xxx.xxx.xxx -> PBXIP/RouterIP

 

restart PBX and It should work fine

 

50/50. It can help in case when static IP used. Most of ISP's changes external IP addresses of home users (DSL) once a day. I need to seek this addresses....

 

This run fine on my side

 

BTW you should have a very very very good Internet connection if you want to keep audio quality for all external registration on your PBX otherwise it will be more like a cell phone quality, forcing codec might help.

 

G.711 used only for tests. Sure we will use G.729 in future.

 

hope it help!

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I need to set routers which supports DMZ on all user's locations. Not best way. SIP ALG?

 

No just on the PBX side, on the client side it should be working with no change at all, If needed only you just have to route phone IP with 5060-5061 and RDT ports. better to use static IP for the phone!

 

 

50/50. It can help in case when static IP used. Most of ISP's changes external IP addresses of home users (DSL) once a day. I need to seek this addresses....

 

I use no-ip.com dynamic DNS it prevent loosing adresse everyday

and I use PBX IP/DNS name

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SIP messages reaching ip phones.

I checked that home router do not blocking the traffic from pbx. Pbxnsip send media traffic to internal IP address, and this media can't be delivered to the ip phone. This is the problem.

 

Sorry, I can't attach the trace "Upload failed. You are not permitted to upload this type of file".

 

You can send the trace to support@pbxnsip.com

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I sent the trace already to support@pbxnsip.com

 

That trace does not contain ACK for the 200 OK message. That's why I think that the phone did not get any message from the PBX. Since you are using snom phone here, you can look at the SIP trace and make sure that the phone is getting messages from the PBX. You can also run the pcap trace on the snom phone (debugging purposes only)

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