August 2005 - sr-users - kamailio.org

[Serusers] NAT traversal solution by reinvite?
by Federico Giannici 01 Sep '05

01 Sep '05

I'd like to ask to somebody with more knowledge of me if a possible solution to NAT traversal is really feasible. For various reasons, we DON'T want to use an RTP proxy. We'd like to avoid the use of STUN because: 1) creates hairpin problems; 2) many UAC have a bad STUN client code implementation; 3) it requires additional configuration by the final user. It seems to me that with the nathelper's message rewriting functions it is possible to solve every problem for the SIP protocol. Moreover, as we have the REAL IP of the UA (in the original SIP messages) we could also avoid haipin problems: it is sufficient to use the original IP/Port of the two UAs if both have the same natted IP (ie they are behind the same NAT). This doesn't work when the UAs are behind multiple NATs, but this is a relatively uncommon case. So, the unresolved problem is with the RTP data, because we don't know what will be the NATted port so we cannot correctly mangle the SDP data in the INVITE message. Am I correct up to this point? Now, I'm asking myself if it is feasible to use a "MINI RTP Proxy" that receives the initial INVITEs, discovering the NATted RTP ports, and then IMMEDIATLY RE-INVITE the two UAs to connect directly each other. So only the first RTP packet is actually proxed, all subsequent traffic will be directly between the two UAs. I think that something similar is done by Asterisk. Is this feasible? If it is, then we could have a good solution to NAT Traversal: 1) No Hairpin problems (for one NAT cases) 2) No problems of the normal RTP proxy (waste of bandwidth, longer delays, bad scalability). 3) Will work with all type of NATs except for symmetric ones (the same that work with STUN). 4) Simpler UAC configuraton: only username, password and sip server. Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Serusers] ip based auth and billing
by Iqbal 01 Sep '05

01 Sep '05

Hi Has anyone attempted this, to see if it works, where the UA is using IP address only, Iqbal

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[Serusers] forward breaks URI rewrite?
by Chris Roberts 01 Sep '05

01 Sep '05

Heya, This might be an obvious answer, but when I forward a packet to another box using the forward() method, this seems to break the URI? rewriteuri("sip:test@test.co.uk"); forward(x.x.x.x, 5060); Now on my asterisk box and looking at ngrep, the test(a)test.co.uk has disappeared, and asterisk takes the SIPDOMAIN as 'x.x.x.x', rather than 'test.com'. Is this the desired behaviour? Is there a way to do what I want to do? I need to be able to pick out the original SIPDOMAIN in asterisk even after the packet has been forwarded by SER. People seem to have talked about doing this, but I'm not sure how I can forward it without obliterating the URI. Cheers, Chris. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.16/83 - Release Date: 26/08/2005

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[Serusers] xmlrpc calls
by zkeatts 01 Sep '05

01 Sep '05

I am currently trying to do an xml-rpc call to usrloc.show_contacts. The xml packet I am sending is the following POST /RPC2 HTTP/1.0 User-Agent: Radio UserLand/7.1b7 (WinNT) Host: localhost:5060 Content-Type: text/xml Content-length: 131 <?xml version=\"1.0\"?> <methodCall> <methodName>usrloc.show_contacts</methodName> <params> </params> </methodCall> As a return packet I am getting HTTP/1.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:32769 Server: Sip EXpress router (0.10.99-janakj_experimental (i386/linux)) Content-Length: 303 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=5820 req_src_ip=127.0.0.1 req_src_port=32769 in_uri=/RPC2 out_uri=/RPC2 via_cnt==1" <?xml version="1.0" encoding="UTF-8"?> <methodResponse> <fault> <value><struct> <member><name>faultCode</name> <value><i4>400</i4></value></member> <member><name>faultString</name> <value><string>More Parameters Expected</string></value></member> </struct></value> </fault> </methodResponse> This tells me that I am not entering the correct information, but I am at a loss to exactly what else I should be inserting. I did a search for rpc_show_contacts and found in ./sip_router/modules/usrloc/ul_rpc.c static void rpc_show_contacts(rpc_t* rpc) After searching some more I could not find the rpc_t structure. Does anyone know what values I should be passing via XML? Thanks, Zach Keatts Software Engineer Nuvio.com

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[Serusers] URGENT:Authenitcation Mode
by Ashutosh Kumar 01 Sep '05

01 Sep '05

Hi, IF I set auth_db:calculate_ha1=no, will the radius authentication also be in non-digest (plaintext/basic) mode?? Reply please. Very urgent. Regards, Ashutosh Kumar

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[Serusers] maxfwd module with Max-Forwards > 255 broken
by Hendrik Scholz 01 Sep '05

01 Sep '05

Hi! I checked how SER would behave with large Max-Forwards values. Paragraph 20.22 of RFC 3261 notes Max-Forwards to be anywhere between 0 and 255 but SER lacks a check for the upper boundry in the maxfwd module. I'm not sure how the correct way to remove the broken header field and introducing a correct one would be so I leave this for somebody who knows :) Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

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[Serusers] overflowing Expire values
by Hendrik Scholz 01 Sep '05

01 Sep '05

Hi! I gave 3.1.2.4 of draft-ietf-sipping-torture-tests-07 a try and sent - a REGISTER with Expires > 2^31 and lower than 2^32 (2147483748 in my case). - a REGISTER with Expires > 2^32 (4294967596 = 2^32 + 300) I used print_expires() to print the resulting values SER (0.9.3 + changes) parsed. It's a negative value in first case and 300 in the latter one. parser/parse_expires.h defines exp_body_t to have 'int val' which is not enough to store the parsed values. Changing it to 'unsigned int val' prevents the negative values from coming up in SER but does not solve the main problem. The draft stats that a default value of 3600 seconds should be assumed in case of errors, so an additional check to make sure _e->val will always be below 2^32 is necessary. I do have a small patch for our non-standard branch and could wrap one up for HEAD if needed. Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

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[Serusers] Blocking calls from unregistered users
by rpagquil＠philonline.com 31 Aug '05

31 Aug '05

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[Users] DROP statement
by Bogdan-Andrei Iancu 31 Aug '05

31 Aug '05

Hi there, The core "drop" statement has a new special meaning in some route types (by default it's equivalent to "exit"): - BRANCH ROUTE - the request branch is discarded (not fwd); - ONREPLY ROUTE - if provisional reply (<200), it will be discarded (not fwd, not stored, but only timers updated); Wiki will be updated soon with this text ;) regards, bogdan

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[Serusers] RURI or URI
by Nando 31 Aug '05

31 Aug '05

hello, somebody knows the true difference between this: sip:1111@sip.domain.com y sip:1111 at sip.domain.com Because in logs (with ngrep) only I see the URI default????? (sip:1111@.....) Why i do not see the RURI??? Thanks

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