Re: [OpenSIPS-Devel] [opensips] B2B GUIDs too short (#783)
Bogdan, Thank you, will do so ASAP! --- You are receiving this because you are subscribed to this thread. Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/issues/783#issuecomment-199514912___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [opensips] B2B GUIDs too short (#783)
In `b2b_entities/dlg.c:b2b_generate_key()`, the use of this format: `..` leads to very short Call-IDs like this: `Call-ID: B2B.27572.17705` This is impractically short, and is certain to lead to collisions in a high-volume environment with millions of calls daily. There are many database systems etc. that rely on all calls being identifiable by a unique GUID. I don't think it conforms to the RFC 3261 prescription that GUIDs be good GUIDs. Unfortunately, it's not possible to simply append additional random data to the key string, since the Call-ID has specific meaning that is extracted in sequential requests, as per `b2b_entities/dlg.c:b2b_parse_key()`. This function also foresees the length of the GUID and the positioning of the delimiters to be rather static in nature. Some sort of solution to increase overall Call-ID length and combinatorial complexity is needed, such as appending time-based data into the string. Much appreciated! --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/issues/783___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [B2B_ENTITIES] Difference between sending CANCEL and BYE
BYEs are sequential (in-dialog) requests that go to the RURI that is the Contact returned in the final 2xx reply. CANCELs are sent to the proximate endpoint - same RURI as initial INVITE request. They are not sequential requests, are not loose- routed, etc. So, the behaviour you are seeing is normal. -- Sent from mobile device On Jan 22, 2010, at 12:55 PM, Olivier Détour chino540off+kamai...@gmail.co m wrote: Hi, I would like to break a SIP communication during a call or a proceeding SIP session... If the call is in progress (Caller can speak to Callee), I can send a BYE to caller and callee each with send_request function; Here is my Wireshark trace: BYE sip:s...@1.1.12.3 SIP/2.0 Via: SIP/2.0/UDP 2.2.22.2;branch=z9hG4bKda73.5742b3a7.0 From: sip:1...@2.2.22.2;tag=934c0604001f8a49c065a1707fbb682d-59cf To: sip:2...@1.1.12.3;tag=B2B.333.0.1264181639 CSeq: 4 BYE Call-ID: B2B.269.0.1264181639 Content-Length: 0 User-Agent: OpenSIPS (1.6.1-notls (i386/linux)) Contact: sip:s...@2.2.22.2 If the sip session is proceeding (Caller send INVITE but he hasn't receive 200 OK yet), I send a CANCEL like I sent the BYE (send_request); But here is my Wireshark trace: CANCEL sip:2...@1.1.12.3 SIP/2.0 Via: SIP/2.0/UDP 2.2.22.2;branch=z9hG4bK5161.0ab21263.0 From: sip:1...@2.2.22.2;tag=934c0604001f8a49c065a1707fbb682d-ac1f To: sip:2...@1.1.12.3 CSeq: 2 CANCEL Call-ID: B2B.134.0.1264181129 User-Agent: OpenSIPS (1.6.1-notls (i386/linux)) Contact: sip:s...@2.2.22.2 Why is the Request Line not for sip:s...@1.1.12.3 as for the BYE? Regards, -- Olivier Détour Sent from Paris, Île-de-France, France ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [OpenSIPS-Users] new CDRTool release 6.9.0
ram wrote: Hi Adrian I found some problem last version when iam patching freeradius.patch with Freeradius 2.1.6 is that Fixed in this version I have installed Freeradius 2.0.4 I would wager, on Adrian's behalf, that a description of some problem or a reference to a bug report number would be necessary to answer this question. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Developer Docs?
The Kamailio folks have a development guide: http://www.kamailio.org/docs/openser-devel-guide/ Much of what is said there applies to OpenSIPS as well - at least, so far. A great deal of it is generally true of the OpenSER code stock. Andaleeb Roomy wrote: Hello, I am new to OpenSIPS. My target is to make a SIP Application Server for presence and messaging according to GSM RCS initiative. I did some exploration of the presence and rls features from the website. I think I will have to extend these to meet OMA/RCS specs. I am quite lost when I peek into the code to understand it. As it is written in plain C, not object oriented C++. Can anyone point me to some sort of development guide or doc that shows how different modules interact with each other, how to extend a module, what basics does a module need, how can one module be used from another, what the general flow of execution is with core and modules, overview of the architecture of OpenSIPS, how to parse messages, the commonly used codes etc. I think it would have been helpful to have some document from the devel course link, but I could not find any. Thanks, Andaleeb ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
