sr-users November 2007

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138 participants
249 discussions

[OpenSER-Users] FYI: [Fwd: [Sip] SIPit 21 summary]
by Klaus Darilion 30 Nov '07

30 Nov '07

-------- Original-Nachricht -------- Betreff: [Sip] SIPit 21 summary Datum: Mon, 19 Nov 2007 14:55:53 -0600 Von: Robert Sparks <rjsparks(a)nostrum.com> An: sip List <sip(a)ietf.org> SIPit 21 was hosted by the BII Group and the Beijing University of Posts and Telecommunications, November 5-9 in Beijing, China. The event was a little smaller than the most recent SIPits: There were 94 attendees from 38 companies visiting from 15 countries. There were 44 teams and around 70 distinct implementations. The majority of the testing I saw was focusing on advanced scenarios rather than basic registration and call setup. As with SIPit 20, the most common area of interoperability problems centered around offer/answer, particularly when attempting to negotiate alternate versions of streams or to explicitly state parameters to be used with a given stream. Forty of the attending teams completed the survey - from those answers: The roles represented (some implementations act in more than one role): 21 endpoints 10 proxy/registrars 3 standalone registrars 5 event servers 4 gateways 6 automaton UAs (voicemail, conference, appserver, etc) 8 b2bua/sbcs 4 UAs with signaling, but without media 1 test/monitoring tool Implementations using each transport for SIP messages: UDP 100% TCP 82% TLS 49% (server auth only) TLS 6% (mutual auth) SCTP 3% DTLS 0% 36% of the implementations supported SIP over IPv6 (up from 25% at SIPit20) 18% supported SIP over IPSec For DNS we had support for: Full RFC3263 : 61% SRV only : 21% A records only : 9% no DNS support : 9% Support for various items: 61% rport 15% sigcomp 22% enum 21% sending multiplexing STUN and SIP 28% receiving multiplexed STUN and SIP 22% RFC4320 fixes 12% identity 70% session-timer There was one implementation of parts of the session-policy framework. There was one implementation of the sip-consent framework. There were two implementations of parts of the sip-config framework (I do not know yet if they worked together). There were three implementations of outbound and four of GRUU. There were two implementations of MSRP present. The endpoints implemented these methods: 100% INVITE, CANCEL, ACK, BYE 87% REGISTER 87% OPTIONS 87% NOTIFY <- still a lot of unsolicted NOTIFYs 87% REFER 77% SUBSCRIBE 77% INFO 70% UPDATE 67% PRACK 47% MESSAGE 33% PUBLISH Support for various extensions in the endpoints: 43% RFC3323 privacy 41% RFC3327 path 13% RFC3840 pref 13% RFC3841 caller-prefs 59% RFC3891 replaces 20% RFC4488 norefersub 0% RFC4538 target-dialog <- There were several folks starting to talk about supporting this Support for various things in the endpoints: 10% ICE (but there was no interoperability - see below) 0% ICE-TCP 13% STUNbis 17% TURN (again, there was no interoperability) 75% symmetric RTP 25% SRTP 0% RTP over DTLS This is how the endpoints answered how they supported multipart/mime: 7% I break if someone sends me multipart/mime 30% I pretend multipart mime doesn't exist if someone sends it to me 20% I ignore multipart/mime, but will proxy it or hand it to my application if it shows up 17% I try to do something useful with multipart/mime I receive, but I never send it 3% I ignore multipart/mime I receive, but I try to send it to do something useful 20% I try to do something useful with multipart/mime I send and receive There were 4 endpoints that would send media over TCP - none of them supported RFC4145 comedia. One of those supported media over TLS. Implementation is all over the map for fork-loop-fix. However, only 3 of the proxy/b2buas present were still vulnerable to the simple form of the attack described in the draft. There were no implementations present with support for location- conveyance or the geopriv common-policy framework. There was one implementation present with support for the RFC4967 dial-string parameter, and one with support for the ecrit service-urn. Of the SIP-Events implementations, the following packages were supported: 62% refer 48% message-summary 38% presence 29% dialog 19% reg 19% conf There was one implementation of each of the following packages: ua-profile certificate/credentials vx-rtcpxp (sipping-rtcp-summary) There was only one implementation of event-lists present. Only 20% of the SIP-Events implementations supported winfo. We had a multiparty sesssion for a full morning focusing on NAT traversal. Basic use of STUN with SIP is hightly interoperable. No two implementations of TURN could even start trying to talk to each other (each having implemented to different points in the recent torrent of changes). I don't think we'll get much implementation feedback until the spec stops making big changes so frequently. No two implementations of ICE worked together. The closest was between two implementations that have worked in the past, but failed during this session when the connectivity checks arrived before the SDP. _______________________________________________ Sip mailing list https://www1.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implementors(a)cs.columbia.edu for questions on current sip Use sipping(a)ietf.org for new developments on the application of sip

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[Serusers] Segmentation Fault (SER Not Starting)
by Alexandr Volkov 29 Nov '07

29 Nov '07

Hi All, I am having trouble when trying to start SER. I am using Ser 0.9.6 on centos4. When I am trying to start it gives: Segmentation Fault; but only if in 'location' table there is a row with callId starting by number: <field name="callid">49c72d7ed685500e7d844d1cf1357dbd</field> if I change this field in database (simple add any character), ser will start. <field name="callid">a49c72d7ed685500e7d844d1cf1357dbd</field> seems like a bug

1 0

[Serusers] ERROR: forward_reply: no 2nd via found in reply - via parsing error
by Tomasz Zieleniewski 29 Nov '07

29 Nov '07

Hi, I have a chcek via parameter set in the ser.cfg file When ser receives reply with such via header it returns ERROR: forward_reply: no 2nd via found in reply This Via seems to be ok? Via: SIP/2.0/UDP 192.168.0.165:7160;branch=z9hG4bK1589.3bc310b3.1,SIP/2.0/UDP 192.168.0.165:7060;branch=z9hG4bK1589.e1306596.0,SIP/2.0/UDP 192.168.129.74 ;rport=5060;branch=z9hG4bK1589.606a3e61.0,SIP/2.0/UDP 192.168.129.74:5160 ;branch=z9hG4bK1589.a5bd9fb4.0,SIP/2.0/UDP 192.168.129.74;branch= z9hG4bK1589.506a3e61.0,SIP/2.0/UDP 192.168.0.102:7060 ;rport=7060;branch=z9hG4bK-d87543-a228576fa32c0b22-1--d87543- What is wrong? Cheers tomasz

1 0

[Serusers] Unexpected disconnect - lost SIP packets?
by Jeremy A 29 Nov '07

29 Nov '07

Hello, I'm need some help with SIP protocol / SER configuration to resolve a problem. I have a system with a SER instance and a SIP client application running on the same box. SER is 10.50.32.10:5060 APP is 10.50.32.10:5070 Remote SIP server 10.1.39.10 Remote phone is 10.1.39.202 The user agent is registered with the SER instance as extension 1009. It has aliases 1001-1008 and is actually called as 231x which maps onto 100x which then connects to extension 1009. The remote phone makes a call to the user agent application via the remote SIP server and the SER instance. It dials 2311 which gets automagically converted to extension 1009 The call is established but the user application times out and detects a disconnect despite the calling phone not hanging up. In my wireshark trace it seems that SER is ignoring some form of keep-alive from the remote SIP server and this is causing the timeout. The lines of interest include 39, 42, 45, 50, 53, 56, 59 Lines 28 and others similar are on the loopback interface and are packets between port 5060 and 5070 Can someone please advise if the SIP protocol has problems and/or if SER configuration has problems? 22 0.218826 10.50.32.10 -> 127.0.0.1 SIP Request: REGISTER sip:localhost.localdomain 23 0.219570 127.0.0.1 -> 10.50.32.10 SIP Status: 200 OK (1 bindings) 24 19.472555 10.1.39.10 -> 10.50.32.10 SIP/SDP Request: INVITE sip:1001@butler.fesa.sto;transport=udp, with session description 25 19.473394 10.50.32.10 -> 10.1.39.10 SIP Status: 100 trying -- your call is important to us 26 19.473596 10.50.32.10 -> 10.50.32.10 SIP/SDP Request: INVITE sip:1009@10.50.32.10:5070;LINEID=1d5a67e38809, with session description 27 19.484244 10.50.32.10 -> 10.50.32.10 SIP Status: 100 Trying 28 19.725476 10.50.32.10 -> 10.50.32.10 SIP Status: 180 Ringing 29 19.725828 10.50.32.10 -> 10.1.39.10 SIP Status: 180 Ringing 32 21.770798 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 33 21.782437 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 38 21.789396 10.50.32.10 -> 10.1.39.202 RTCP Source port: 8001 Destination port: 16391 39 21.800069 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 40 22.278096 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 41 22.278713 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 42 22.298101 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 43 23.285536 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 44 23.288988 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 45 23.307839 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 48 25.298214 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 49 25.299105 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 50 25.317892 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 51 29.306496 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 52 29.306758 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 53 29.328420 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 54 33.314736 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 55 33.318736 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 56 33.338302 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 57 37.323175 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 58 37.324791 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 59 37.347936 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 --> client detects disconnect here Thanks jeremy

3 8

[Serusers] Presence in SIP
by Carlos Angeles 29 Nov '07

29 Nov '07

Hello I have installed Ser-2.0 with the PA module.... I am testing the server with x-lite... When a user is NOTIFYING some event change the server is answering with a Status: 420 Bad Extension.... Is thst SER doesn´t support this kind of presence events? Thanks Carlos _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

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[OpenSER-Users] Handling re-INVITES --hellp required
by srinivas Antarvedi 29 Nov '07

29 Nov '07

Hello all, i have users one is on global ip and another behind NAT am using asterisk as media server leg 1: caller : Global ip UAC. callee: asterisk leg2: caller :asterisk callee: NATed UAC. sdp of NATed client is handled at openser reply route at first stage when asterisk re-invites the NATed UAC to bridge the two call-Leg's the sdp from NATed UAC is not changed ,, even if i call t_on_reply in the loose route section of the script.. it is still showing privat ip so finally after 2 or 3 sec's there was an end to the dialog can anybody have any idea to handle re-invite's 200 ok SDP mangling? please help me out.. Thanks in advance regards srinivas -- Srinivas Antarvedi

3 3

[OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Mihail Gluhowchenko 29 Nov '07

29 Nov '07

Hello All; I`m trying to tune openser-1.2.2 with dbtext data base But have some problems. I make db use - textdb.sh Then add users use - sc.dbtext and add to openser.cfg: #set module path mpath="/usr/local/lib/openser/modules/" # use dbtext database loadmodule "dbtext.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # use dbtext database for persistent storage modparam("usrloc", "db_mode", 2) modparam("usrloc|auth_db", "db_url", "dbtext:///usr/local/etc/openser/dbtext") # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "domain_column", "domain") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); log(1, "Too Many Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); log(1, "Message too big\n"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { log(1, "this is a request REGISTER users\n"); if (!www_authorize("my.domain", "subscriber")) { log(1, "this is a www_challenge\n"); www_challenge("my.domain", "0"); exit; }; save("location"); exit; }; but when trying REGISTER user use(sjphone) in log i see: Nov 28 17:30:03 devel s-eltex[12797]: this is a request for my.domain users Nov 28 17:30:03 devel s-eltex[12797]: this is a request REGISTER users Nov 28 17:30:03 devel s-eltex[12797]: this is a www_challenge Nov 28 17:30:03 devel s-eltex[12794]: this is a request for my.domain users Nov 28 17:30:03 devel s-eltex[12794]: this is a request REGISTER users Nov 28 17:30:03 devel s-eltex[12794]: get_ha1(): Error while querying database In sjphone LOG i see: Service unavailable server error; What do I adjust not correctly?

2 1

[OpenSER-Users] t_relay can not handle PRACK retransmission
by Klaus Darilion 29 Nov '07

29 Nov '07

Hi! I have encountered this problem at an openser 1.1.1 proxy: Client Proxy Client ---PRACK------> ------PRACK-----> retansmission -----PRACK-------> here at this point (when openser receives the PRACK retransmission) t_relay fails: ERROR: t_newtran: transaction already in process 0xb66e7880 Is this a known problem? Were there some fixes to 1.1 or 1.2 branch? regards klaus

2 4

[OpenSER-Users] Branch limitations
by Daniel 29 Nov '07

29 Nov '07

Hi all, I've read this from iptel website. It is true for SER 0.11.0. What about Openser 1.2 ? " * Script processing of multiple branches on forking Warning ser's request processing language allows to make request decisions based on current URI. When a request if forked to multiple destinations, only the first branch's URI is used as input for script processing. This might lead to unexpected results. Whenever a URI resolves to multiple different next-hop URIs, only the first is processed which may result in handling not appropriate for the other branch. For example, a URI might resolve to an IP phone SIP address and PSTN gateway SIP address. If the IP phone address is the first, then script execution ignores the second branch. If a script includes checking gateway address in request URI, the checks never match. That might result in ignoring of gateway admission control rules or applying them unnecessarily to non-gateway destinations. List of known problems is publicly available at the ser webpage at [70]http://www.iptel.org/ser/ . See the "ISSUES" link. -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] OpenSER: Support of advanced RegExpr?
by Helmut Kuper 28 Nov '07

28 Nov '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is openser supporting advanced regular expressions as described here: http://www.regular-expressions.info/refadv.html regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTVg64tZeNddg3dwRAl+JAKCst6W4QdNjJ6ZT6fWRaE9otPcX+wCdEKbw iQy1J1Md3CKE7pUgx4lld3k= =sRw7 -----END PGP SIGNATURE-----

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sr-users November 2007