Audio is being stripped from RTP stream from VoIP Provider
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The edge gateway has a LAN IP of 192.168.1.254 /24 and the pfSense has a WAN IP of 192.168.1.1 /24. The LAN side of the pfSense is 172.16.1.254 /24.
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Ah so your doing nat.. And would not be able to prove that that edge router is not messing with the packets ;)
What is that device is it PE or CE? So your saying you can not sniff on it? Can you put a tap between your wan and its wan and sniff there? Or a switch with span port works, or a old school hub works too if your speeds are not to high..
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Just a couple questions.
1. What makes you think just the audio is being stripped?
2. Is it being stripped in both directions?A troubleshooting possibility would be to place a phone or phones between the edge gateway and the pfSense box.
Joe
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Apparently, NAT is quite difficult for VoIP to deal with and it appears you're doing it twice. Voip-info.org is where I go for information and resources for my asterisk PBXs and other VoIP specific info and resources. Something that might help get you going in the right direction is a post on their site that talks about VoIP wth NAT issues.
http://www.voip-info.org/wiki/view/NAT+and+VOIP
Hope that helps.
Joe
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I am indeed NATing yes, and I am NATing twice… I know NAT can be problematic with VoIP but there is no reason why this shouldn't work. I suppose I could enable outbound static ports on the pfSense and then technically I am only NATing on the edge gateway.
This is all set up in VMware vCloud Director so I have no control of the edge gateway at all, and the functionality is somewhat limited from the portal that I see. I guess the edge gateway would fall into the PE category, however I suspect this is somewhat virtualised and does not physically exist at all.
In response to Joe - We are basically sending the audio out to the VoIP provider from the PBX, we see this leave the PBX and we are not receiving any audio in from the VoIP provider from the caller. I cannot confirm or deny whether or not the audio leaves the WAN port outgoing or reaches the WAN incoming which is proving to be my stumbling block at the minute.
One other thing to note is that when the caller rang, they reported this as a "dead call" as in they didn't even hear the ringing at their side. Now I'm not sure where the 180 ringing originates from but maybe this could provide an insight into something.
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Also just to add to this, if this WAS a NAT issue would I still be receiving the RTP stream? I still receive the RTP stream when I have no sound on the call. In the last instance, I recieved 1293 RTP packets from the SIP provider, however the payload was all d5's and there was no sound at all. Surely if this was a NAT issue the RTP stream would not be reaching the PBX at all?
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That's what I'd expect as well but I understand that some of the NAT issues occur even when NAT appears to be working fine. Not to say that's definitely the reason for your problem but apparently you can get mysterious behaviour. Generally speaking there isn't anything I can think of that once set would cause an intermittent issue as infrequent as what you're describing.
Since you say that you have the PBX internally and that's terminated by another company then it's quite possible that you're not actually using SIP across the NAT and we can rule out most of those issues. Asterisk servers are generally setup to use IAX2 which is NAT friendly. However, a one way call is a NAT symptom. The article below seems to explain in more detail how NAT and more specifically PAT can affect your call. Perhaps it will spark some ideas.
https://sipsorcery.wordpress.com/2009/08/05/nat-rtp-and-audio-problems/
Good luck
Joe
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"I suppose I could enable outbound static ports on the pfSense and then technically I am only NATing on the edge gateway."
No you would still be natting twice you just would not be changing the ports twice with napt. which is how most nat works.. You end up with this, say your talking to a http site
privateIP:52111 –- host-publicIP:80 (nat router) router-publicIP:53222 --- host-publicIP:80
What you have is this
privateIP:52111 --- host-publicIP:80 (nat router) difprivateIP:53222 --- host-publicIP:80 (2nd nat router) router-publicIP:50333 ---- host-publicIP:80
Changing to use static outbound ports does not remove your double with 2 different private IPs you just remove the port number changes.. Which if trying to use static could cause even more issues, when that port is already in use by some other connection when you have multiple hosts that all pick their source port on their own.. You have no way to tell machine X don't use ports X - Y because we want machine B to use them. And now you told the router he can not change the port on the public side to be one that is not in use in the state table because you set it to be static.
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Also just to add to this, if this WAS a NAT issue would I still be receiving the RTP stream? I still receive the RTP stream when I have no sound on the call. In the last instance, I recieved 1293 RTP packets from the SIP provider, however the payload was all d5's and there was no sound at all. Surely if this was a NAT issue the RTP stream would not be reaching the PBX at all?
RTP is media and SIP is signalling. You can receive the RTP media but your client only listens the IP-port pair which is sent by the SDP in SIP.
So basically, your client may listen NAT IP where as media(RTP) has destination of your WAN IP (or vice versa)
btw, why don't you use a soft SIP client such as linphone and capture the packets from that PC.
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Voip was not originally built with the idea of NAT. It was a commercial endeavor and not originally marketed at the residential side of things. NAT was added in later.
Below is information my voip provider shows about one of my lines and how its connected. Notice the NATted address is included in the "Contact" line. There is no room for a second NATted address. I can't say it wont work but I don't know how you would make it do so. You might consider using Siproxd package on your pfSense box. It will take out one of the NATs for you. Or at least fool things into thinking that way.
Received:sip:1x4x.2x.2x.x1:5060
Contact:sip:36xxxxxxx9@__172.25.125.20__:5060;user=phone -
"I suppose I could enable outbound static ports on the pfSense and then technically I am only NATing on the edge gateway."
No you would still be natting twice you just would not be changing the ports twice with napt. which is how most nat works.. You end up with this, say your talking to a http site
privateIP:52111 –- host-publicIP:80 (nat router) router-publicIP:53222 --- host-publicIP:80
What you have is this
privateIP:52111 --- host-publicIP:80 (nat router) difprivateIP:53222 --- host-publicIP:80 (2nd nat router) router-publicIP:50333 ---- host-publicIP:80
Changing to use static outbound ports does not remove your double with 2 different private IPs you just remove the port number changes.. Which if trying to use static could cause even more issues, when that port is already in use by some other connection when you have multiple hosts that all pick their source port on their own.. You have no way to tell machine X don't use ports X - Y because we want machine B to use them. And now you told the router he can not change the port on the public side to be one that is not in use in the state table because you set it to be static.
This is very true, I was over looking NAT in this instance. I think for the time being I will leave NAT and outbound NAT as it is as I don't want to make things worse than they already are!
Voip was not originally built with the idea of NAT. It was a commercial endeavor and not originally marketed at the residential side of things. NAT was added in later.
Below is information my voip provider shows about one of my lines and how its connected. Notice the NATted address is included in the "Contact" line. There is no room for a second NATted address. I can't say it wont work but I don't know how you would make it do so. You might consider using Siproxd package on your pfSense box. It will take out one of the NATs for you. Or at least fool things into thinking that way.
Received:sip:1x4x.2x.2x.x1:5060
Contact:sip:36xxxxxxx9@__172.25.125.20__:5060;user=phoneI have read about the Siproxd package, but I am not entirely sure exactly how this works. I will need to do a bit more investigation about this. Thanks for all the advice - it has given me something to go on! :)