SIP Connections

[DanCarroll]

Hello all

I have a SIP machine here in Connecticut.

I have a number of 3rd party SIP phones in INDIA starting to connect to it.

I can call them and they can call me. I can hear and so can they.

But when 1 India extension calls another they get 1 way audio.

As the SIP signalling is correct and they get connected, my questions  are--

Does the connection at that point become from softphone to softphone on the remote LAN?

Could my firewall and/or routers here in CT. have anything to do with this?

Any sugestions would be appreciated

[bosoxfan_1]

yes, when India softphone speaks to India softphone, the connection is on the local LAN and the audio travels from softphone to softphone, not to the NBX in CT.

We have had issues with one way audio when there is a firewall in between when setting up the call. Check your firewall settings and make sure it is not SIP enabled.... that has caused some 1 way audio issues.

[DanCarroll]

The firewalls are SIP enabled.

I can make a call to India and it works. They can call me in CT and it works.

I had 2 seperate calls going at the same time to India with no issues,so the RTP port number assignment is correct.

It just is when  they call themselves in India that the issue occurs.

[bosoxfan_1]

yes, like I said, when a firewall is SIP enabled, i have seen the firewall change the SIP message to a phone so the phone sends audio to the gateway instead. You would need a trace to see where the audio is going.

Not sure how many firewalls inbetween but you should try disabling SIP.

I know the firewall that had the issue was a Fortigate firewall

[DanCarroll]

Just to recap..

I did the traces

Ext 3966 in India calls ext 3965 in India.

3966 softphone  is 10.xxx.25.88

3965 softphone is 10.xxx.25.89

TRACE FROM 3966------------

Invite goes  to 3965  with port 40020  RTP/AVP 0 8 3 101.

200 ok with sdp replies with same codec and port 40000

RTP trace at this point  from 3966

shows ----

source is 10.xxx.25.88 port40020, to Destination 10.xxx.25.89 port 40000

CORRECT.

But --No reverse RTP packets at all.

TRACE FROM 3965

Invite recieved  3965@10.130.50.17:22936

SIP/SDP sends back 60698 RTP/AVP 0 8 3 101

RTP scans show-

Source is 10.xxx.25.88 port 40020 to Destination 10.xxx.25 89 port 40000

No reverse of 10.xxx.25.89 port 40000  to 25.88. port 40020

BUT have

Sourse is 10.130.25.89 port 40000 to 10.130.50.17port 60698

So the called extension can hear, but not respond.

This seems to indicate that 2 invites were being sent to ext 3965 for  the same call.

The calling extension(3966) got the correct info, (correct IP's and ports) to set up the call.

The called extension could hear but not transmit.

Is this still a firewall issue????

Any help would be appreciated.

Thanks

[bosoxfan_1]

yes, it seems that it is a firewall issue to me..

What is IP address 10.230.50.17? is that a firewall?

We have had issues where some firewalls act as application layer gateway and change the SIP packet that the NBX sends out and points to a proxy... Disable SIP on the firewall and everything should work fine.

[merlin_1]

Hi Dan

Did you disable sip on the firewall ?

[DanCarroll]

I can put the request in.

 But whose firewall will become sip disabled?

Should I wait until you review the traces?