[OpenSIPS-Devel] [ opensips-Patches-2706135 ] Change to listen= syntax to support per-interface advertised
Phil D'Amore
ptdamore at gmail.com
Tue Aug 25 19:08:24 CEST 2009
If you have UAs behind your NAT why not just have them register to a
different IP address on the OpenSIPS box, and don't specify an
advertised address for that binding? The registrar will automatically
select this interface when you do a lookup() for one of those UAs.
Double Record-Route headers are properly added by OpenSIPS when a
message enters one interface and leaves another. That worked even
before this patch, which is why you do not see it explicitly mentioned
here. One of the things the patch does is make sure that the
advertised address (if specified) winds up on the Record-Route header
corresponding to the interface with the alias.
Thanks,
Phil
> Date: Sun, 23 Aug 2009 22:33:43 +0200
> From: Emmanuel BUU <emmanuel.buu at ives.fr>
> Subject: Re: [OpenSIPS-Devel] [ opensips-Patches-2706135 ] Change to
> listen= syntax to support per-interface advertised
> To: OpenSIPS devel mailling list <devel at lists.opensips.org>
> Cc: noreply at sourceforge.net
> Message-ID: <4A91A7A7.4090300 at ives.fr>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello,
>
> This is a good idea except that the real live situation is more complicated.
>
> Advertised addresses are used whenether OpenSIPS is running behind an
> external NAT (provided by a router). If fact, the opensips may have to
> deal with UAs that are also behind the NAT (e.g. inside a corporate
> network) where there is no need to advertise other address the bind
> address. So, we need to distinguish between:
>
> - messages remaining inside an "internal network"
> - messages going through the NAT
>
> To be able to make this distinction, we would need to have some
> description of what the Internal network is lile
>
> internal_network=10.0.0.0/8
>
> If any outgoing message matches these networks, the advertised address
> would not be used. It would be great to be able to have more than one
> such internal network description.
>
> And for advertised address, there is another issue: loose routing. In
> case of routing a packet from an internal network to the external
> network, we should have a double record route (like if we had two
> interfaces)
>
> Record-Route: sip:bind_address:port
> Record-Route: sip:adv_address:port
>
> Tricky ... but asterisk does it nicely.
>
>
> SourceForge.net a ?crit :
>> Patches item #2706135, was opened at 2009-03-23 15:50
>> Message generated for change (Settings changed) made by bogdan_iancu
>> You can respond by visiting:
>> https://sourceforge.net/tracker/?func=detail&atid=1086412&aid=2706135&group_id=232389
>>
>> Please note that this message will contain a full copy of the comment thread,
>> including the initial issue submission, for this request,
>> not just the latest update.
>> Category: core
>> Group: trunk
>>
>>> Status: Closed
>>> Resolution: Accepted
>>>
>> Priority: 5
>> Private: No
>> Submitted By: Phil D'Amore (ptdamore)
>> Assigned to: Bogdan-Andrei Iancu (bogdan_iancu)
>> Summary: Change to listen= syntax to support per-interface advertised
>>
>> Initial Comment:
>> [This is the patch for an idea I posted several weeks ago to the devel list.
>> Below is the original list posting as an explanation.]
>>
>> The patch extends the idea of the advertised_address and advertised_port
>> parameters so that it is possible to bind such an address to each
>> listen= statement.
>>
>> Syntax-wise, it works like this:
>>
>> listen=[proto:]host[:port][ AS host[:port]]
>>
>> This new AS host[:port] syntax binds the advertised address specifically to the
>> corresponding socket_info structure. If such an address is specified, it will
>> be used before the address specified as advertised_address, or set in the
>> set_advertised_address function for packets being sent on that particular
>> socket. If it is not specified, then the old methods work as before.
>>
>> The core has been modified so that address tuples using the advertised address
>> work interchangeably with the real address where you'd expect them to.
>> grep_sock_info and find_si have been updated to allow this. Therefore, it is
>> not necessary to explicitly name your advertised address on an alias= line.
>> The following are now equivalent:
>>
>> force_send_socket(udp:adv_address:adv_port);
>> force_send_socket(udp:real_address:real_port);
>>
>> Note we don't specify protocol on the advertised address. The way I see
>> things, it makes no sense to advertise a different protocol, so it inherits the
>> protocol of the socket.
>>
>> Via construction is affected in the same way that the old global
>> advertised_address does, except the Via is written specifically for
>> the socket used
>> to send the message.
>>
>> Lump substitution is also affected. The following lump substitutions are aware
>> of this new facility:
>>
>> SUBST_RCV_IP
>> SUBST_RCV_PORT
>> SUBST_RCV_ALL
>> SUBST_SND_IP
>> SUBST_SND_PORT
>> SUBST_SND_ALL
>>
>> As a result of this, I was able to stop using record_route_preset in
>> my own config, and
>> just use standard record_route. This winds up working much better,
>> because double-rr
>> works properly, and add_rr_param now works as expected for me, where it was a
>> problem using record_route_preset.
>>
>> Additionally, registrar and usrloc had to be modified:
>>
>> registrar - update the add_sock_hdr function to put the advertised address in
>> the message instead of the real address. This allows registration information
>> to be replicated between multi-homed proxies, and the correct interface on the
>> peer can be used.
>>
>> usrloc - update db_insert_ucontact to store the advertised address in the
>> database instead of the real address. Same reasoning as the mods to registrar.
>>
>> ----------------------------------------------------------------------
>>
>>
>>> Comment By: Bogdan-Andrei Iancu (bogdan_iancu)
>>>
>> Date: 2009-08-23 00:44
>>
>> Message:
>> Hi Phil,
>>
>> The patch is on SVN - thank for your contribution.
>>
>> Best regards,
>> Bogdan
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Phil D'Amore (ptdamore)
>> Date: 2009-07-13 19:47
>>
>> Message:
>> Bogdan,
>>
>> Yes the listen_alias_port_fixed.patch file contains a complete patch with
>> the known issues fixed.
>>
>> Looking forward to your assessment.
>>
>> Thanks,
>> Phil
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Bogdan-Andrei Iancu (bogdan_iancu)
>> Date: 2009-07-13 14:36
>>
>> Message:
>> Hi Phil,
>>
>> Is the listen_alias_port_fixed.patch the final version with all the bugs
>> fixed ? If so, I will review the code and upload it on SVN.
>>
>> Thanks and regards,
>> Bogdan
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Phil D'Amore (ptdamore)
>> Date: 2009-07-10 21:05
>>
>> Message:
>> Hi Bogdan:
>>
>> Any word if this will be included in the next release now that the bug is
>> fixed?
>>
>> Thanks,
>> Phil
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Nergal (nergalex)
>> Date: 2009-06-03 13:07
>>
>> Message:
>> Hello Phil,
>>
>> That's great, the bug is fixed!
>> Results of no regression tests is OK too.
>>
>> Thank you for your help.
>> Alexis
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Phil D'Amore (ptdamore)
>> Date: 2009-05-26 20:03
>>
>> Message:
>> Alexis:
>>
>> Thanks for the extra information. I think I found the cause of the issue,
>> and I've uploaded a new version of the patch. Please try the
>> listen_alias_port_fixed.patch. I have not had time to test the issue in my
>> environment since I use port 5060 everywhere. Since I don't know if I'll
>> be able to do it today I'm uploading the modified patch so you can test if
>> you'd like.
>>
>> The issue was in the changes to msg_translator.c. I was incorrectly using
>> the length of the original port to calculate an offset. It was not
>> properly re-written with the rest of the code to use the alias port when
>> it's available.
>>
>> Thanks,
>> Phil
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Bogdan-Andrei Iancu (bogdan_iancu)
>> Date: 2009-05-26 19:52
>>
>> Message:
>> Hi Guys,
>>
>> Is the bug fixed in the patch ? I just want to know to start evaluating
>> the patch for upload.
>>
>> Thanks and regards,
>> Bogdan
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Nergal (nergalex)
>> Date: 2009-05-26 12:44
>>
>> Message:
>> Hello Phil,
>>
>> After few tests, I identified the exact context when the problem appear. I
>> answer to your questions before.
>>
>> 1) b) An extra random character, this morning is '.' that appears in the
>> first
>> Record-Route header after host:port BUT always in a specific context, see
>> #3.
>>
>> 2) I have "listen=X.X.X.X:5060 AS X.X.X.Y:5060" in my configuration and
>> also:
>> listen=X.X.X.X:15060 AS X.X.X.Y:5060
>>
>> 3) [NEW] The problem appears when the SIP message match the
>> "X.X.X.X:15060" instance. The rewrite of RR is done by the line
>> "listen=X.X.X.X:15060 AS X.X.X.Y:5060". In this case, the extra caracter is
>> suffixed to the first RR port (public inbound interface).
>> The problem appears for UDP and TCP protocol.
>>
>> NB: 2 instances (5060 and 15060) of my SIP proxy is load-balanced. So, the
>> random effect is made by:
>> - If "SIP message destination = 5060" So test OK
>> - If "SIP message destination = 15060" So test Not OK
>>
>> Best regards
>> Alexis
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Phil D'Amore (ptdamore)
>> Date: 2009-05-25 23:18
>>
>> Message:
>> Alexis :
>>
>> I have a few questions for you:
>>
>> 1) To make sure I understand the problem, I want to re-word it a bit.
>> Which of these statements is true about the problem
>> a) An extra ';' (always a ';') character appears in the first
>> Record-Route header after host:port, at random times.
>> b) An extra random character (not always ';') appears in the first
>> Record-Route header after host:port, at random times
>>
>> 2) Since you are using double-rr can I assume that you also have a
>> listen=X.X.X.X:5060 line in your configuration for your public inbound
>> interface, but it was not included in your message because it does not
>> specify the [AS host:port] portion?
>>
>> If the answer to 2 is yes, can you try to specify an alias on that listen=
>> line like this:
>>
>> listen=X.X.X.X:5060 AS X.X.X.X:5060
>>
>> The effect should be that no re-write appears to happen as the left and
>> right side are the same, but the code used to get there would be slightly
>> different. I'm curious to see if the extra ';' goes away in that case.
>>
>> Thanks,
>> Phil
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Nergal (nergalex)
>> Date: 2009-05-25 11:20
>>
>> Message:
>> Hello,
>>
>> Sorry to post this comment so late, I was in vacancy.
>> History: https://sourceforge.net/forum/message.php?msg_id=7379817
>>
>> Best regards
>> Alexis
>> -------------------
>> I have a bug on the patch "Change to listen= syntax to support
>> per-interface advertised - ID: 2706135". In the first Record-Route header,
>> a random caracter ends the <IP:Port> entry (see below 2 tests with the same
>> configuration and the extra caracter is ';').
>> Could you help me to fix it?
>> I can do more test or give you more information if you need.
>>
>> Maybe a clue: the Advertised port is 4 digits long but the real port is 5
>> digits long.
>>
>> conf extract:
>> - X.X.X.X is a public IP
>> - listen=tcp:10.143.136.8:15060 AS 10.143.136.29:5060
>> - force_send_socket(10.143.136.8:15060);
>> - t_relay("tcp:10.143.81.130:5060")
>>
>>
>> --- Test OK / proto TCP ---
>> 1. Before SIP proxy processing (Front interface capture)
>> SUBSCRIBE sip:service at pilote-sip.my_domain:5060 SIP/2.0
>> Via: SIP/2.0/TCP 10.95.10.1:10005;branch=z9hG4bK-1338-1-0
>> From: userA at sfr.fr <sip:sipp at 10.95.10.1:10005>;tag=1338SIPpTag001
>> To: userA at sfr.fr <sip:service at pilote-sip.my_domain:5060>
>> Call-ID: 1-1338 at 10.95.10.1
>> CSeq: 1 SUBSCRIBE
>> Contact: sip:sipp at 10.95.10.1:10005
>> Max-Forwards: 70
>> Subject: Subscribe Test
>> Content-Length: 0
>>
>> 2. After SIP proxy processing (Back interface capture)
>> SUBSCRIBE sip:service at pilote-sip.my_domain:5060 SIP/2.0
>> Record-Route:
>> <sip:10.143.136.29:5060;transport=tcp;r2=on;lr=on;ftag=1338SIPpTag001>
>> Record-Route:
>> <sip:X.X.X.X:5060;transport=tcp;r2=on;lr=on;ftag=1338SIPpTag001>
>> Via: SIP/2.0/TCP 10.143.136.29:5060;branch=z9hG4bKa4b3.1d7725a.0;i=9e8f
>> Via: SIP/2.0/TCP
>> 10.95.10.1:10005;rport=32769;received=10.95.10.1;branch=z9hG4bK-1338-1-0
>> From: userA at sfr.fr <sip:sipp at 10.95.10.1:10005>;tag=1338SIPpTag001
>> To: userA at sfr.fr <sip:service at pilote-sip.my_domain:5060>
>> Call-ID: 1-1338 at 10.95.10.1
>> CSeq: 1 SUBSCRIBE
>> Contact: sip:sipp at 10.95.10.1:10005
>> Max-Forwards: 69
>> Subject: Subscribe Test
>> Content-Length: 0
>> --- end Test OK ---
>>
>> --- Test NOK / proto TCP ---
>> 1. Before SIP proxy processing (Front interface capture)
>> SUBSCRIBE sip:service at pilote-sip.my_domain:5060 SIP/2.0
>> Via: SIP/2.0/TCP 10.95.10.1:10003;branch=z9hG4bK-1571-1-0
>> From: userA at sfr.fr <sip:sipp at 10.95.10.1:10003>;tag=1571SIPpTag001
>> To: userA at sfr.fr <sip:service at pilote-sip.my_domain:5060>
>> Call-ID: 1-1571 at 10.95.10.1
>> CSeq: 1 SUBSCRIBE
>> Contact: sip:sipp at 10.95.10.1:10003
>> Max-Forwards: 70
>> Subject: Subscribe Test
>> Content-Length: 0
>>
>> 2. After SIP proxy processing (Back interface capture)
>> SUBSCRIBE sip:service at pilote-sip.my_domain:5060 SIP/2.0
>> Record-Route:
>> <sip:10.143.136.29:5060;transport=tcp;r2=on;lr=on;ftag=1571SIPpTag001>
>> Record-Route:
>> <sip:X.X.X.X:5060;;transport=tcp;r2=on;lr=on;ftag=1571SIPpTag001>
>> Via: SIP/2.0/TCP 10.143.136.29:5060;branch=z9hG4bK60ec.650af5e3.0;i=83401
>> Via: SIP/2.0/TCP
>> 10.95.10.1:10003;rport=32808;received=10.95.10.1;branch=z9hG4bK-1571-1-0
>> From: userA at sfr.fr <sip:sipp at 10.95.10.1:10003>;tag=1571SIPpTag001
>> To: userA at sfr.fr <sip:service at pilote-sip.my_domain:5060>
>> Call-ID: 1-1571 at 10.95.10.1
>> CSeq: 1 SUBSCRIBE
>> Contact: sip:sipp at 10.95.10.1:10003
>> Max-Forwards: 69
>> Subject: Subscribe Test
>> Content-Length: 0
>> --- end Test NOK ---
>>
>> Best regards
>> Alexis
>>
>> ----------------------------------------------------------------------
>>
>> Comment By: Bogdan-Andrei Iancu (bogdan_iancu)
>> Date: 2009-05-09 11:15
>>
>> Message:
>> Hi Phil,
>>
>> I will start reviewing the patch in order to upload it on SVN. If I have
>> some questions about it, I will post them here.
>>
>> Thanks and regards,
>> Bogdan
>>
>> ----------------------------------------------------------------------
>>
>> You can respond by visiting:
>> https://sourceforge.net/tracker/?func=detail&atid=1086412&aid=2706135&group_id=232389
>>
>> _______________________________________________
>> Devel mailing list
>> Devel at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>>
>>
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