May be I miss some important details? No suggestions?
Thank you.
Hello, all.
Using nathelper + rtpproxy for subj. Kamailio has public and private network interfaces. Asterisk is only private. RTP Proxy is working in bridge mode and relaying traffic from UAC to Asterisks.
Everything is working fine, except one configuration. When the client is behind router ( a specific one, I do not have an access there to check ), and this UAC is making a call to other public extension, which is behind router, then RTP Proxy is relaying traffic to the caller, using another UDP port, then the packets arrive.
For instance: UAC 1 -> UAC 2
PUBLIC_IP:10 > KAMAILIO_IP:5555 KAMAILIO_IP:5678 > PUBLIC_IP:12
While for the UAC 2 it looks like:
PUBLIC_IP:20 > KAMAILIO_IP:6767 KAMAILIO_IP:4564 > PUBLIC_IP:20
The source and destination UDP ports are the same. As result, I can hear UAC 1 and he cannot hear me.
In case of we have UAC 3, which is behind other router, call is working fine with same configuration.
"It's routers fault" you can say, but in the same configuration ( I mean network, not kamailio ) it worked, but when RTPProxy was not in bridge mode and Kamailio and Asterisks were in public network. Reinvites are not allowed in both cases.
The question is, why the source and destination UDP ports are different? Using STUN in first case, cause without it, private IP written in contacts and as result, traffic relayed from Kamailio is incorrect, cause heading to private network which is unreachable.
Any ideas where to dig?
Hello,
one option might be a bad ALG implementation in the router.
Can you send a full ngrep of such case? You can obfuscate the IP addresses, use different ones for each point in the network and leave the ports. Seeing SIP headers and SDP can indicate the presence of an ALG or something broken in config logic.
Also, what is the parameter you give to force_rtp_proxy(...)?
Cheers, Daniel
On 3/2/11 8:38 AM, Spinov Evgeniy wrote:
May be I miss some important details? No suggestions?
Thank you.
Hello, all. Using nathelper + rtpproxy for subj. Kamailio has public and private network interfaces. Asterisk is only private. RTP Proxy is working in bridge mode and relaying traffic from UAC to Asterisks. Everything is working fine, except one configuration. When the client is behind router ( a specific one, I do not have an access there to check ), and this UAC is making a call to other public extension, which is behind router, then RTP Proxy is relaying traffic to the caller, using another UDP port, then the packets arrive. For instance: UAC 1 -> UAC 2 PUBLIC_IP:10> KAMAILIO_IP:5555 KAMAILIO_IP:5678> PUBLIC_IP:12 While for the UAC 2 it looks like: PUBLIC_IP:20> KAMAILIO_IP:6767 KAMAILIO_IP:4564> PUBLIC_IP:20 The source and destination UDP ports are the same. As result, I can hear UAC 1 and he cannot hear me. In case of we have UAC 3, which is behind other router, call is working fine with same configuration. "It's routers fault" you can say, but in the same configuration ( I mean network, not kamailio ) it worked, but when RTPProxy was not in bridge mode and Kamailio and Asterisks were in public network. Reinvites are not allowed in both cases. The question is, why the source and destination UDP ports are different? Using STUN in first case, cause without it, private IP written in contacts and as result, traffic relayed from Kamailio is incorrect, cause heading to private network which is unreachable. Any ideas where to dig?
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Unfortunately ngrep is unavailable right now, cause network was configured to use public IPs. May be I'll can do that on development network later. Right now development network using public`s also.
I'll try to sort out ngrep anyway.
I was giving FAEI to INVITEs from UAC to Asterisk and FAIE to INVITEs from Asterisks to UAC. Everything was good except destination UDP port to UAC 1. It was different then the source. As result UAC 1 didn't received backflow.
Also, may be this will help: Kamailio was unable to identify that faulty UAC 1 is behind the NAT. I've tried nat_uac_test("31"), however - nothing, while SIP headers were containing NATed IPs. So during tests I've just forced NAT always. Without that I didn't had audio at all. While with it - one way audio with faulty UAC and normal call for all others.
Also, on faulty UAC 1 I had to use STUN server, while all other clients worked without it. After going Asterisks public and changing kamailio configuration for it, STUN no longer needed anywhere.
Just assuming fact, that router has bad ALG implementation. Is there any workaround for it, may be forcing destination ports to source ones?
On Wed, 2011-03-02 at 09:30 +0100, Daniel-Constantin Mierla wrote:
Hello,
one option might be a bad ALG implementation in the router.
Can you send a full ngrep of such case? You can obfuscate the IP addresses, use different ones for each point in the network and leave the ports. Seeing SIP headers and SDP can indicate the presence of an ALG or something broken in config logic.
Also, what is the parameter you give to force_rtp_proxy(...)?
Cheers, Daniel
On 3/2/11 8:38 AM, Spinov Evgeniy wrote:
May be I miss some important details? No suggestions?
Thank you.
Hello, all. Using nathelper + rtpproxy for subj. Kamailio has public and private network interfaces. Asterisk is only private. RTP Proxy is working in bridge mode and relaying traffic from UAC to Asterisks. Everything is working fine, except one configuration. When the client is behind router ( a specific one, I do not have an access there to check ), and this UAC is making a call to other public extension, which is behind router, then RTP Proxy is relaying traffic to the caller, using another UDP port, then the packets arrive. For instance: UAC 1 -> UAC 2 PUBLIC_IP:10> KAMAILIO_IP:5555 KAMAILIO_IP:5678> PUBLIC_IP:12 While for the UAC 2 it looks like: PUBLIC_IP:20> KAMAILIO_IP:6767 KAMAILIO_IP:4564> PUBLIC_IP:20 The source and destination UDP ports are the same. As result, I can hear UAC 1 and he cannot hear me. In case of we have UAC 3, which is behind other router, call is working fine with same configuration. "It's routers fault" you can say, but in the same configuration ( I mean network, not kamailio ) it worked, but when RTPProxy was not in bridge mode and Kamailio and Asterisks were in public network. Reinvites are not allowed in both cases. The question is, why the source and destination UDP ports are different? Using STUN in first case, cause without it, private IP written in contacts and as result, traffic relayed from Kamailio is incorrect, cause heading to private network which is unreachable. Any ideas where to dig?
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 3/2/11 9:32 AM, Spinov Evgeniy wrote:
Unfortunately ngrep is unavailable right now, cause network was configured to use public IPs. May be I'll can do that on development network later. Right now development network using public`s also.
I'll try to sort out ngrep anyway.
I was giving FAEI to INVITEs from UAC to Asterisk and FAIE to INVITEs from Asterisks to UAC. Everything was good except destination UDP port to UAC 1. It was different then the source. As result UAC 1 didn't received backflow.
You say about wrong port for RTP or for SIP?
For SIP be sure you call force_rport(). For RTP try eventually the flag 'r' in in parameters of force_rtp_proxy().
Also, may be this will help: Kamailio was unable to identify that faulty UAC 1 is behind the NAT. I've tried nat_uac_test("31"), however - nothing, while SIP headers were containing NATed IPs.
By NATed ip you mean private class, like 10... or 192.168...? If yes, that is strange, can you add debugger module with cfgtrace enabled to see what lines in the config file are executed for that call? (this is assuming you are using v3.1.x, if not add xlog() messages in the config to be sure the nat handling part is executed).
Cheers, Daniel
So during tests I've just forced NAT always. Without that I didn't had audio at all. While with it - one way audio with faulty UAC and normal call for all others.
Also, on faulty UAC 1 I had to use STUN server, while all other clients worked without it. After going Asterisks public and changing kamailio configuration for it, STUN no longer needed anywhere.
Just assuming fact, that router has bad ALG implementation. Is there any workaround for it, may be forcing destination ports to source ones?
On Wed, 2011-03-02 at 09:30 +0100, Daniel-Constantin Mierla wrote:
Hello,
one option might be a bad ALG implementation in the router.
Can you send a full ngrep of such case? You can obfuscate the IP addresses, use different ones for each point in the network and leave the ports. Seeing SIP headers and SDP can indicate the presence of an ALG or something broken in config logic.
Also, what is the parameter you give to force_rtp_proxy(...)?
Cheers, Daniel
On 3/2/11 8:38 AM, Spinov Evgeniy wrote:
May be I miss some important details? No suggestions?
Thank you.
Hello, all. Using nathelper + rtpproxy for subj. Kamailio has public and private network interfaces. Asterisk is only private. RTP Proxy is working in bridge mode and relaying traffic from UAC to Asterisks. Everything is working fine, except one configuration. When the client is behind router ( a specific one, I do not have an access there to check ), and this UAC is making a call to other public extension, which is behind router, then RTP Proxy is relaying traffic to the caller, using another UDP port, then the packets arrive. For instance: UAC 1 -> UAC 2 PUBLIC_IP:10> KAMAILIO_IP:5555 KAMAILIO_IP:5678> PUBLIC_IP:12 While for the UAC 2 it looks like: PUBLIC_IP:20> KAMAILIO_IP:6767 KAMAILIO_IP:4564> PUBLIC_IP:20 The source and destination UDP ports are the same. As result, I can hear UAC 1 and he cannot hear me. In case of we have UAC 3, which is behind other router, call is working fine with same configuration. "It's routers fault" you can say, but in the same configuration ( I mean network, not kamailio ) it worked, but when RTPProxy was not in bridge mode and Kamailio and Asterisks were in public network. Reinvites are not allowed in both cases. The question is, why the source and destination UDP ports are different? Using STUN in first case, cause without it, private IP written in contacts and as result, traffic relayed from Kamailio is incorrect, cause heading to private network which is unreachable. Any ideas where to dig?
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On Wed, 2011-03-02 at 09:53 +0100, Daniel-Constantin Mierla wrote:
On 3/2/11 9:32 AM, Spinov Evgeniy wrote:
Unfortunately ngrep is unavailable right now, cause network was configured to use public IPs. May be I'll can do that on development network later. Right now development network using public`s also.
I'll try to sort out ngrep anyway.
I was giving FAEI to INVITEs from UAC to Asterisk and FAIE to INVITEs from Asterisks to UAC. Everything was good except destination UDP port to UAC 1. It was different then the source. As result UAC 1 didn't received backflow.
You say about wrong port for RTP or for SIP?
On RTP only. SIP works fine, except STUN server requirement as I've mentioned below.
For SIP be sure you call force_rport(). For RTP try eventually the flag 'r' in in parameters of force_rtp_proxy().
According to README, r flag affects IP addresses, so it will not solve RTP issue, but I'll try it to get rid of STUN server requirement, hope this will help.
Also, may be this will help: Kamailio was unable to identify that faulty UAC 1 is behind the NAT. I've tried nat_uac_test("31"), however - nothing, while SIP headers were containing NATed IPs.
By NATed ip you mean private class, like 10... or 192.168...?
Yea, in my case this is 192.168....
If yes, that is strange, can you add debugger module with cfgtrace enabled to see what lines in the config file are executed for that call? (this is assuming you are using v3.1.x, if not add xlog() messages in the config to be sure the nat handling part is executed).
This is Kamailio 3.0.4. I've added xlogs and seen that messages are proceeded on SIP sessions ( ones for INVITE from UAC1 and once for INVITE from Asterisk to UAC2 ) and for RTP, once for each session. RTP Proxy is proxing all 4 flows ( 2 per each side ). May be I should take a look at something else?
Cheers, Daniel
So during tests I've just forced NAT always. Without that I didn't had audio at all. While with it - one way audio with faulty UAC and normal call for all others.
Also, on faulty UAC 1 I had to use STUN server, while all other clients worked without it. After going Asterisks public and changing kamailio configuration for it, STUN no longer needed anywhere.
Just assuming fact, that router has bad ALG implementation. Is there any workaround for it, may be forcing destination ports to source ones?
On Wed, 2011-03-02 at 09:30 +0100, Daniel-Constantin Mierla wrote:
Hello,
one option might be a bad ALG implementation in the router.
Can you send a full ngrep of such case? You can obfuscate the IP addresses, use different ones for each point in the network and leave the ports. Seeing SIP headers and SDP can indicate the presence of an ALG or something broken in config logic.
Also, what is the parameter you give to force_rtp_proxy(...)?
Cheers, Daniel
On 3/2/11 8:38 AM, Spinov Evgeniy wrote:
May be I miss some important details? No suggestions?
Thank you.
Hello, all. Using nathelper + rtpproxy for subj. Kamailio has public and private network interfaces. Asterisk is only private. RTP Proxy is working in bridge mode and relaying traffic from UAC to Asterisks. Everything is working fine, except one configuration. When the client is behind router ( a specific one, I do not have an access there to check ), and this UAC is making a call to other public extension, which is behind router, then RTP Proxy is relaying traffic to the caller, using another UDP port, then the packets arrive. For instance: UAC 1 -> UAC 2 PUBLIC_IP:10> KAMAILIO_IP:5555 KAMAILIO_IP:5678> PUBLIC_IP:12 While for the UAC 2 it looks like: PUBLIC_IP:20> KAMAILIO_IP:6767 KAMAILIO_IP:4564> PUBLIC_IP:20 The source and destination UDP ports are the same. As result, I can hear UAC 1 and he cannot hear me. In case of we have UAC 3, which is behind other router, call is working fine with same configuration. "It's routers fault" you can say, but in the same configuration ( I mean network, not kamailio ) it worked, but when RTPProxy was not in bridge mode and Kamailio and Asterisks were in public network. Reinvites are not allowed in both cases. The question is, why the source and destination UDP ports are different? Using STUN in first case, cause without it, private IP written in contacts and as result, traffic relayed from Kamailio is incorrect, cause heading to private network which is unreachable. Any ideas where to dig?
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
So, I've made resetup of network to run tests.
I've figured out, that one of the routers were changing SIP headers, i.e. Contacts header to public IP. This caused UAC under this router not to use STUN and gave one way audio.
Now I'm using fix_nated_contact and fix_nated_register. Location table contains correct IPs. However, when I'm making a call, RTP is sent to private IP. With STUN IP is correct. Where it comes from?
I tried also STUN and force_rtp_proxy() with 'r' flag. Nothing has changed - destination port for RTP traffic differs from source port. As result - complete silence.
Please advice, where to dig?
On Wed, 2011-03-02 at 09:53 +0100, Daniel-Constantin Mierla wrote:
On 3/2/11 9:32 AM, Spinov Evgeniy wrote:
Unfortunately ngrep is unavailable right now, cause network was configured to use public IPs. May be I'll can do that on development network later. Right now development network using public`s also.
I'll try to sort out ngrep anyway.
I was giving FAEI to INVITEs from UAC to Asterisk and FAIE to INVITEs from Asterisks to UAC. Everything was good except destination UDP port to UAC 1. It was different then the source. As result UAC 1 didn't received backflow.
You say about wrong port for RTP or for SIP?
For SIP be sure you call force_rport(). For RTP try eventually the flag 'r' in in parameters of force_rtp_proxy().
Also, may be this will help: Kamailio was unable to identify that faulty UAC 1 is behind the NAT. I've tried nat_uac_test("31"), however - nothing, while SIP headers were containing NATed IPs.
By NATed ip you mean private class, like 10... or 192.168...? If yes, that is strange, can you add debugger module with cfgtrace enabled to see what lines in the config file are executed for that call? (this is assuming you are using v3.1.x, if not add xlog() messages in the config to be sure the nat handling part is executed).
Cheers, Daniel
So during tests I've just forced NAT always. Without that I didn't had audio at all. While with it - one way audio with faulty UAC and normal call for all others.
Also, on faulty UAC 1 I had to use STUN server, while all other clients worked without it. After going Asterisks public and changing kamailio configuration for it, STUN no longer needed anywhere.
Just assuming fact, that router has bad ALG implementation. Is there any workaround for it, may be forcing destination ports to source ones?
On Wed, 2011-03-02 at 09:30 +0100, Daniel-Constantin Mierla wrote:
Hello,
one option might be a bad ALG implementation in the router.
Can you send a full ngrep of such case? You can obfuscate the IP addresses, use different ones for each point in the network and leave the ports. Seeing SIP headers and SDP can indicate the presence of an ALG or something broken in config logic.
Also, what is the parameter you give to force_rtp_proxy(...)?
Cheers, Daniel
On 3/2/11 8:38 AM, Spinov Evgeniy wrote:
May be I miss some important details? No suggestions?
Thank you.
Hello, all. Using nathelper + rtpproxy for subj. Kamailio has public and private network interfaces. Asterisk is only private. RTP Proxy is working in bridge mode and relaying traffic from UAC to Asterisks. Everything is working fine, except one configuration. When the client is behind router ( a specific one, I do not have an access there to check ), and this UAC is making a call to other public extension, which is behind router, then RTP Proxy is relaying traffic to the caller, using another UDP port, then the packets arrive. For instance: UAC 1 -> UAC 2 PUBLIC_IP:10> KAMAILIO_IP:5555 KAMAILIO_IP:5678> PUBLIC_IP:12 While for the UAC 2 it looks like: PUBLIC_IP:20> KAMAILIO_IP:6767 KAMAILIO_IP:4564> PUBLIC_IP:20 The source and destination UDP ports are the same. As result, I can hear UAC 1 and he cannot hear me. In case of we have UAC 3, which is behind other router, call is working fine with same configuration. "It's routers fault" you can say, but in the same configuration ( I mean network, not kamailio ) it worked, but when RTPProxy was not in bridge mode and Kamailio and Asterisks were in public network. Reinvites are not allowed in both cases. The question is, why the source and destination UDP ports are different? Using STUN in first case, cause without it, private IP written in contacts and as result, traffic relayed from Kamailio is incorrect, cause heading to private network which is unreachable. Any ideas where to dig?
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Daniel-Constantin Mierla -
Spinov Evgeniy