Hello. I have a FS behind firewall on which ports 5080,5060 udp are mapped to the FS server. A client connected directly to Internet, say using GPRS, authorise to FS and calling a number audio is heared. When this client is connected via WiFi to some router connected directly to Internet, client authorises, calls but there is no audio. So I'm stump on where to look for issue: on FS server side with firewall or on client side behind router? Thank you. -- Mimiko desu.
It seems like it is associated with either the portforwarding of the wifi router or NAT prolem, I guess. The following link could be of help, imho: http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT /zenny On 12/10/12, Mimiko <vbvbrj@gmail.com> wrote:
Hello.
I have a FS behind firewall on which ports 5080,5060 udp are mapped to the FS server. A client connected directly to Internet, say using GPRS, authorise to FS and calling a number audio is heared. When this client is connected via WiFi to some router connected directly to Internet, client authorises, calls but there is no audio.
So I'm stump on where to look for issue: on FS server side with firewall or on client side behind router?
Thank you. -- Mimiko desu.
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On 10.12.2012 16:58, Zenny wrote:
It seems like it is associated with either the portforwarding of the wifi router or NAT prolem, I guess. The following link could be of help, imho: http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT
I've read all pages on wiki related to nat and still have trouble finding the cause. -- Mimiko desu.
So to clarify: Client -> Internet -> NAT -> FS works Client -> NAT -> Internet -> NAT -> FS does not? Since the FS NAT side appears to be working (from the 1st case), I'm guessing it's probably either a problem at the client side (with client or NAT router) or perhaps the double-NAT is causing an issue. Personally I haven't run FS behind NAT, so it's not a situation I've had to handle. Do a packet trace of a call at both the client and FS ends. The FS end is easy - "sofia global siptrace on if you have low volumes, or a tool like tcpdump/ngrep if you have higher volumes. FS will actually log the SDP information in at debug-level console logs and in XML CDRs. At the client side try using Wireshark on the client PC. Compare the SDP information of the call, to check the IP and ports seen at either end. Check whether they look reasonable. FS will need to see the public IP of the client, and the client see the public IP of FS for audio to work in both directions. The FS/client may not need to know their public IP on the internal side of the router as many routers have SIP ALG that will rewrite the internal IP to an external one, but if the client has already done so that often causes more problems than it solves. FS can handle only knowing the internal IP of the client if they're able to send RTP to FS - it will see the IP/port the audio is coming from and send audio back to that address. That does mean audio won't work on such a call until it has received RTP from the client though. -Steve On 10 December 2012 13:59, Mimiko <vbvbrj@gmail.com> wrote:
Hello.
I have a FS behind firewall on which ports 5080,5060 udp are mapped to the FS server. A client connected directly to Internet, say using GPRS, authorise to FS and calling a number audio is heared. When this client is connected via WiFi to some router connected directly to Internet, client authorises, calls but there is no audio.
So I'm stump on where to look for issue: on FS server side with firewall or on client side behind router?
Thank you. -- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On 10.12.2012 20:00, Steven Ayre wrote:
So to clarify:
Client -> Internet -> NAT -> FS works
Client -> NAT -> Internet -> NAT -> FS does not?
Since the FS NAT side appears to be working (from the 1st case), I'm guessing it's probably either a problem at the client side (with client or NAT router) or perhaps the double-NAT is causing an issue. Personally I haven't run FS behind NAT, so it's not a situation I've had to handle.
Thank you for a comprehensive response. What I've seen now. Audio actually comes up, but it is taking a while. Sometimes is ~5sec, sometimes more, sometimes never. Also, sip commands does not take action. The other party have to hung up, for example, in order for call to end. -- Mimiko desu.
Do you see any entries in the FS logs relating to rtp auto-adjust at about the time the audio starts working? If so that could indicate FS is getting an incorrect (internal) IP in the SDP from the client, and audio is starting when the client starts sending to FS after a few seconds. The other party have to hung up, for example, in order for call to end. This also indicates a NAT issue. The BYE hangup gets sent to the IP:port in the Contact header, not the address the INVITE came from. If the contact header is incorrect the BYE goes to the wrong place and so the client never sees it. If you can't correct that on the client end (STUN, set manually, SIP ALG etc) then try http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction and/or http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport which will tell FS to break from the SIP standard to support buggy clients. -Steve On 10 December 2012 18:22, Mimiko <vbvbrj@gmail.com> wrote:
On 10.12.2012 20:00, Steven Ayre wrote:
So to clarify:
Client -> Internet -> NAT -> FS works
Client -> NAT -> Internet -> NAT -> FS does not?
Since the FS NAT side appears to be working (from the 1st case), I'm guessing it's probably either a problem at the client side (with client or NAT router) or perhaps the double-NAT is causing an issue. Personally I haven't run FS behind NAT, so it's not a situation I've had to handle.
Thank you for a comprehensive response.
What I've seen now. Audio actually comes up, but it is taking a while. Sometimes is ~5sec, sometimes more, sometimes never. Also, sip commands does not take action. The other party have to hung up, for example, in order for call to end.
-- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Mimiko, I have had hours and hours of problems with NAT, so I can at least tell you what helped me get past them. All of my problems have been in the configuration PHONE > NAT > Internet
FS .
CISCO SPECIFIC ISSUES: I learned through a long tough week with a new customer that CISCO 3xx and 5xx Series have a web setup section that is critical when behind NAT. When logged in as an ADMIN, on the SIP Tab, these NAT Support Parameters should be set to yes: Handle VIA received, Insert VIA received, Substitute VIA Addr, Handle VIA rport, Insert VIA rport, Send Resp To Src Port With those all set I have had pretty good luck with CISCO. GENERIC TO ALL PHONES: Brian and Michael have just worked me through a similar Registration issue with Polycom phones, but I am sure this applies to many others too. The key to understanding that it was a NAT issue was that the phones CSeq (in the Sophia global sipstrace on) did not change from one attempt to the next. It was always "CSeq: 1 REGISTER" and the second one would be "2 REGISTER" with the auth stuff attached. That shows it did not receive the response. In this case I added a parameter and a variable to the USER section of the "directory" on the server. <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <param name="NDLB-force-rport" value="true"/> I am not sure if that fits your case, but maybe it will help others in the future. Sean -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mimiko Sent: Monday, December 10, 2012 8:59 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Natting question. Hello. I have a FS behind firewall on which ports 5080,5060 udp are mapped to the FS server. A client connected directly to Internet, say using GPRS, authorise to FS and calling a number audio is heared. When this client is connected via WiFi to some router connected directly to Internet, client authorises, calls but there is no audio. So I'm stump on where to look for issue: on FS server side with firewall or on client side behind router? Thank you. -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On 10.12.2012 22:00, Sean Devoy wrote:
Hi Mimiko,
I have had hours and hours of problems with NAT, so I can at least tell you what helped me get past them.
All of my problems have been in the configuration PHONE > NAT > Internet
FS .
I've gived up using NAT on FS server. So I put server on public IP, because if FS is after a firewall then there is too much complexity like different cases how a phone can connect: For example, FS firewall is on public ip: IP-PUB1 A phone is registering with this IP-PUB1, but the client can be in different location: 1) client -> local-net -> FS firewall -> NAT -> FS firewall -> local-net -> local-ip of FS. Ie, client and FS is on same network, but client uses FS public natted IP-PUB1. 2) client -> other-local-nat (wifi) -> router -> local-net -> FS firewall -> NAT -> FS firewall -> local-net -> local-ip of FS. Ie client is behind router located in same network as FS and is connecting to FS IP-PUB1. Triple nat. 3) client -> internet -> FS firewall -> local-net -> local-ip of FS. 4) client -> router -> internet -> FS firewall -> local-net -> local-ip of FS. 5) client -> VPN -> local-net -> FS firewall -> NAT -> FS firewall -> local-net -> local-ip of FS. If in all cases except 3 and 5 is using local-ip of FS, then its ok. But to handle or 5 cases with public IP is troublesome. -- Mimiko desu.
participants (4)
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Mimiko
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Sean Devoy
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Steven Ayre
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Zenny