[Serusers] SER with 2 ethernet + PSTN gateway: RTPPROXY bridge mode conf
Hi, I'm using SER with nathelper and rtpproxy as a proxy to guard a PSTN gateway ( calls can only be generated by SIP client and terminated to PSTN, no SIP to SIP or PSTN to SIP calls allowed, REGISTER is not needed ). PSTN gateway and SER eth0 have pubblic internet IP address ( SER eth0 x.x.x.246, PSTN gateway with two interfaces, x.x.x.243 for SIP and x.x.x.244 for RTP ), SER is setted to force rtpproxy for every call and rtpproxy is started in standard mode: this is running fine, every client from internet ( behind a NAT box or not ) is able to place calls. Now the problem is that I need to collect SIP traffic from another non routable network to the PSTN gateway ( es. 10.0.0.0 ) I have connected SER interface eth1 to this network ( ip 10.0.0.1 ) and I am trying to connect to the PSTN from a client in the network 10.0.0.0: SIP signaling seems to be ok, but I'm unable to setup RTP stream. With this configuration it appears that RTPproxy try to connect to PSTN gateway on x.x.x.243 and not x.x.x.244. I have tryied to set RTP proxy in bridge mode ( rtpproxy -l x.x.x.246/10.0.0.1 ) without succes, but documentation is not very clear about this case (I think it has been designed for client to client, not client to PSTNgateway ). ... if (dst_ip == x.x.x.246) { #!! Internet IP address force_rtp_proxy("FI"); t_on_reply("1"); } else if (dst_ip == 10.0.0.1) { #!! Private IP address force_rtp_proxy("FE"); t_on_reply("1"); }; ... Now the question is: what is the correct design to implement this ? RTPproxy in standard mode ? RTPproxy in bridge mode ? ( how ??? ) Mediaproxy ? Any help would be appreciated, many thanks in advance. Fabio
That's a tricky one :-) A few comments: - I believe bridge mode is necessary and thus rtpproxy is what you need (mediaproxy does not have this capability AFAIK) - rtpproxy just waits for the peers to connect, it does not connect rtp to anything, so if you see traffic from the wrong pstn interface, I would look at GW - Just inspect your INVITE and OK SDP to get the options straight: The one going to your 10. network should have the 10.address in SDP, the one to your GW, should have your .246 address - Please note that 0.9.x has a bug (fixed in cvs head). The problem is related to signalling on different listen ports/interfaces: If SER is configured to listen to more than one port, the signalling may not return with the correct src_ip/port (UDP packet, not SIP message). I'm not sure if this holds for listening on different interfaces as well, but I would think so g-) Fabio Macchi wrote:
Hi,
I'm using SER with nathelper and rtpproxy as a proxy to guard a PSTN gateway ( calls can only be generated by SIP client and terminated to PSTN, no SIP to SIP or PSTN to SIP calls allowed, REGISTER is not needed ).
PSTN gateway and SER eth0 have pubblic internet IP address ( SER eth0 x.x.x.246, PSTN gateway with two interfaces, x.x.x.243 for SIP and x.x.x.244 for RTP ), SER is setted to force rtpproxy for every call and rtpproxy is started in standard mode: this is running fine, every client from internet ( behind a NAT box or not ) is able to place calls.
Now the problem is that I need to collect SIP traffic from another non routable network to the PSTN gateway ( es. 10.0.0.0 )
I have connected SER interface eth1 to this network ( ip 10.0.0.1 ) and I am trying to connect to the PSTN from a client in the network 10.0.0.0: SIP signaling seems to be ok, but I'm unable to setup RTP stream.
With this configuration it appears that RTPproxy try to connect to PSTN gateway on x.x.x.243 and not x.x.x.244.
I have tryied to set RTP proxy in bridge mode ( rtpproxy -l x.x.x.246/10.0.0.1 ) without succes, but documentation is not very clear about this case (I think it has been designed for client to client, not client to PSTNgateway ).
.....
if (dst_ip == x.x.x.246) { #!! Internet IP address
force_rtp_proxy("FI");
t_on_reply("1");
} else if (dst_ip == 10.0.0.1) { #!! Private IP address
force_rtp_proxy("FE");
t_on_reply("1");
};
.....
Now the question is: what is the correct design to implement this ?
RTPproxy in standard mode ?
RTPproxy in bridge mode ? ( how ??? )
Mediaproxy ?
Any help would be appreciated, many thanks in advance.
Fabio
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Greger, thanks for you answer, it helped to place order in my confused ideas. About the bug you mentionated, has it been fixed in the cvs stable branch ( rel_0_8_14 , currently I have upgraded to this release ) or I have to install the main unstable branch ? Ive sniffed a call from the SER interface connected to the GW: it appears that the entire SIP negotiation on this side ( I mean source IP address of UDP packet ) is between the GW ip ( x.x.x.243 ) and the SER IP assigned to the other interface (10.0.0.1 ) but, if Im not wrong, I expected a SER UDP source address x.x.x.246 . Is this the problem ? May it depends from the mentionated bug ? After successfully SIP negotiation, I see RTP packet from GW -> rtpProxy ( x.x.x.244 -> x.x.x.246 ) and from client -> rtpproxy ( 10.0.0.2 -> 10.0.0.1 ), but rtpproxy is not bridging packets between the two interfaces. Another question: is there any documentation about the use of force_rtp_proxy options more complete then nathelper README ? That document only mentionate force_rtp_proxy("FI") and force_rtp_proxy("FE") ( hope this is correct in this case ), but in many example I see force_rtp_proxy("FII") or force_rtp_proxy("FIE") and I dont understand the double option. Many thanks again Fabio _____ Da: Greger V. Teigre [mailto:greger@teigre.com] Inviato: mercoledì 12 luglio 2006 8.46 A: Fabio Macchi Cc: serusers@iptel.org Oggetto: Re: [Serusers] SER with 2 ethernet + PSTN gateway: RTPPROXY bridge mode conf That's a tricky one :-) A few comments: - I believe bridge mode is necessary and thus rtpproxy is what you need (mediaproxy does not have this capability AFAIK) - rtpproxy just waits for the peers to connect, it does not connect rtp to anything, so if you see traffic from the wrong pstn interface, I would look at GW - Just inspect your INVITE and OK SDP to get the options straight: The one going to your 10. network should have the 10.address in SDP, the one to your GW, should have your .246 address - Please note that 0.9.x has a bug (fixed in cvs head). The problem is related to signalling on different listen ports/interfaces: If SER is configured to listen to more than one port, the signalling may not return with the correct src_ip/port (UDP packet, not SIP message). I'm not sure if this holds for listening on different interfaces as well, but I would think so g-) Fabio Macchi wrote: Hi, Im using SER with nathelper and rtpproxy as a proxy to guard a PSTN gateway ( calls can only be generated by SIP client and terminated to PSTN, no SIP to SIP or PSTN to SIP calls allowed, REGISTER is not needed ). PSTN gateway and SER eth0 have pubblic internet IP address ( SER eth0 x.x.x.246, PSTN gateway with two interfaces, x.x.x.243 for SIP and x.x.x.244 for RTP ), SER is setted to force rtpproxy for every call and rtpproxy is started in standard mode: this is running fine, every client from internet ( behind a NAT box or not ) is able to place calls. Now the problem is that I need to collect SIP traffic from another non routable network to the PSTN gateway ( es. 10.0.0.0 ) I have connected SER interface eth1 to this network ( ip 10.0.0.1 ) and I am trying to connect to the PSTN from a client in the network 10.0.0.0: SIP signaling seems to be ok, but Im unable to setup RTP stream. With this configuration it appears that RTPproxy try to connect to PSTN gateway on x.x.x.243 and not x.x.x.244. I have tryied to set RTP proxy in bridge mode ( rtpproxy -l x.x.x.246/10.0.0.1 ) without succes, but documentation is not very clear about this case (I think it has been designed for client to client, not client to PSTNgateway ). .. if (dst_ip == x.x.x.246) { #!! Internet IP address force_rtp_proxy("FI"); t_on_reply("1"); } else if (dst_ip == 10.0.0.1) { #!! Private IP address force_rtp_proxy("FE"); t_on_reply("1"); }; .. Now the question is: what is the correct design to implement this ? RTPproxy in standard mode ? RTPproxy in bridge mode ? ( how ??? ) Mediaproxy ? Any help would be appreciated, many thanks in advance. Fabio _____ _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Inline. Fabio Macchi wrote:
Greger, thanks for you answer, it helped to place order in my confused ideas.
Confusion at a higher level, that's what we all are striving for!!
About the bug you mentionated, has it been fixed in the cvs stable branch ( rel_0_8_14 , currently I have upgraded to this release ) or I have to install the main unstable branch ?
Yuo should at least download 0.9.x. ftp://siprouter.onsip.org/pub/gettingstarted/packages/ I think the fix is only in cvs head, but I'm not sure. Does anybody know? I think Jan did the fix.
I've sniffed a call from the SER interface connected to the GW: it appears that the entire SIP negotiation on this side ( I mean source IP address of UDP packet ) is between the GW ip ( x.x.x.243 ) and the SER IP assigned to the other interface (10.0.0.1 ) but, if I'm not wrong, I expected a SER UDP source address x.x.x.246 . Is this the problem ? May it depends from the mentionated bug ?
Well, if SER is running on two interfaces as well (and not only one public IP), then you need to turn on multihoming (mhome=yes) and you should probably turn on double record routing as well (see rr README) There's also another command in 0.9x where you can actually force a given interface. I have never used it. force_send_socket()
After successfully SIP negotiation, I see RTP packet from GW -> rtpProxy ( x.x.x.244 -> x.x.x.246 ) and from client -> rtpproxy ( 10.0.0.2 -> 10.0.0.1 ), but rtpproxy is not bridging packets between the two interfaces.
Another question: is there any documentation about the use of force_rtp_proxy options more complete then nathelper README ?
That document only mentionate force_rtp_proxy("FI") and force_rtp_proxy("FE") ( hope this is correct in this case ), but in many example I see force_rtp_proxy("FII") or force_rtp_proxy("FIE") and I don't understand the double option.
Not that I know of. However, I searched through my archives and found a message with a working bridge scenario. For some reason, I don't think it was put at onsip.org. With the upcoming new iptel.org site, adding how-tos will be much easier for everybody :-) Meanwhile, I'll forward the message to the list. g-)
participants (2)
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Fabio Macchi -
Greger V. Teigre