sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

[Serusers] Carrier-grade framework for SER

by Andreas Granig

Hi guys, I'd like to propose another possibility for a highly-available and scalable system design based on SER: The problems I've encountered for scalable systems are: - Distribution of the user location and alias location among the nodes (user location is based on registrations, alias location comes from web interfaces and is used for call forwarding). - Reloading up to date location tables after breakdown and recovery of a node So I'm just thinking loud about the following provisioning system: - Write a client which fulfills the this demands: - Receive one or more locations from SER via a SER module or from a web application and distribute them to other known clients. Take care of retransmissions if a client isn't reachable or reports a temporary failure. - Receive one or more locations from other clients and write them into the SER FIFO. If writing into the FIFO fails, try to write directly into the database (location-table, alias-table etc.). Report a temporary failure if this also fails. Maybe a centralized server should be used which receives the locations from the clients and distributes them to other clients, so that the nodes just know about the server and nothing about other nodes. This would make integration of new nodes easy. On the other hand, it's another single point of failure, so a decentralized solution should be considered. But that would mean that you've to inform every node about the existence of a new node. The protocol used between the nodes should be simple and fast. So I think SOAP drops out here. Maybe XMLRPC or ICE (http://www.zeroc.com/ice.html) could be used. One might think now why not just use replication on SIP layer, but t_replicate only supports one peer and you've no possibility to get locations on a node while it's down. Replication of other location tables like the alias-table is also not possible. I'd be willing to release these parts as GPL for creating an open framework for carrier-grade SER integration, so any feedback, improvements or flames are highly welcome. Cheers, Andy

18 years, 1 month

[Serusers] Modifying "From" header field of an INVITE message

by scm-j＠nuntius.com

I am relatively new to SER. My SIP service provider requires me to present the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows: From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx> They want it in E.164 format as follows: From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx> I see function only to modify the SIP URI but not the other header or SDP fields. Could someone help me with a code snippet that I can use in ser.cfg to do this conversion? Regards, SCM

18 years, 6 months

[Serusers] ser 0.9.3 crash

by Tulika Pradhan

on my machine, ser 0.9.3 crashes as soon as a UA sends REGISTER message on doing gdb ser corefile i get the following when calling 'bt' (gdb) bt #0 qm_malloc (qm=0x8107e00, size=760) at mem/q_malloc.c:290 #1 0x0806ce0e in receive_msg ( buf=0x80d6560 "REGISTER sip:194.255.1.199:5060 SIP/2.0\r\nFrom: <sip:88515356@194.255.1.199>;tag=ccd4c5cb-13c4-2838-9d1b45-6d9e\r\nTo: <sip:88515356@194.255.1.199>\r\nCall-ID: ccd4c5cb-13c4-2838-9d1b45-7a20\r\nCSeq: 1 REGIS"..., len=437, rcv_info=0xbfed0fe0) at receive.c:92 #2 0x0808421c in udp_rcv_loop () at udp_server.c:458 #3 0x0805ceaf in main_loop () at main.c:1032 #4 0x0805e50b in main (argc=2, argv=0xbfed11a4) at main.c:1568 (gdb) print h $9 = 0 please help, note that a similar setup on another machine works just fine. tulika

18 years, 7 months

[Serusers] ser listening on multiple ports

by Iqbal

Hi Is it possible to get ser to listen on multiple ports at the same time iqbal

18 years, 7 months

RE: [Serusers] msilo, voicemail and serweb

by Alejandro Mellado

Yes, On serweb acount tab message store. I've seen the voicemail plugin code, and this doesn't write on mysql silo table. How I do put voicemail on serweb? Regards Alejandro. El sáb, 23-07-2005 a las 11:46 +0200, harry gaillac escribió: > Hello, > > You mean forward to voicemail on account tab ? > > harry > --- Alejandro Mellado <amellado(a)uct.cl> a écrit : > > > Hi > > > > > > I'm trying to activate the voicemail in serweb. > > I've working msilo with IM successfully. > > > > Somebody Can help me? > > > > > > Regards > > Saluda Atte. > > Alejandro Mellado <amellado(a)uct.cl>. > > Escuela de Informática > > Universidad Católica de Temuco > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com Saluda Atte. Alejandro Mellado <amellado(a)uct.cl>. Escuela de Informática Universidad Católica de Temuco

18 years, 10 months

Re: [Serusers] Distributing "OSP enable SER" binary file w/ source code

by vmathur＠transnexus.com

Hernan, OSP uses Public key based authentication and encrytion schemes, which are stronger than Radius's shared secrets. If you use OSP for authentication and authorization, you need not use Radius. OSP, just like Radius, has a client stack, which is implemented in the SER, and has a Server, which provides centralized call routing, accounting, and security. At the time of startup, the OSP server and the client, the SER in this case, exchange their public keys. The public and private keys can then be used to encode/decode messages as desired. A typical call setup procedure looks like this 1. The source SER goes to the osp server to get the SIP URI corresopnding to the destination. The message is encoded using the source SER's private key. 2. The osp server decodes the message using source SERs public key and after successful decoding (authentication) returns the route back to the source. Along with the route, it also sends back a digitally signed (using the OSP Servers private key) token. 3. The Source uses the route returned by the OSP Srver to send an INVITE. The INVITE message contains the token issued by the OSP Server 4. The destination decodes/validates the token using the OSP Servers public key. Upon successful validation (authorization) it accepts the call. This saves you the effort of mantaining cumbersome access lists for authentication. I can provide you with more documentatin on OSP and how to use SER with OSP if you wish Thanks, Vikrant ------------------------------------------------------------------------------------------------------------- Fogive my ignorance for I have never heard of OSP before ;) You mentioned Radius in your message. How does Radius authentication work in OSP? I am having a tough time getting mine to work. hernan vmathur(a)transnexus.com wrote: Dear All, I have recently implemented OSP w/ SER. OSP is an ETSI defined protocol, which I am using for ceneralised routing, and security of my inter-domain calls. The problem, however, is that the build process is a little lengthy. I want to contribute my implementation to this group so that anyone who is struggling with SER routing configurations or Radius authentication issues may benefit from it. I was, thus, wondering if we can have a binary file of the OSP enabled SER, that can be distributed with the source code. Does anyone have an opinion on this? Also, for anyone who wants to check-out this implementation, I can provide more details. Thanks, Vikrant _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

18 years, 10 months

[Serusers] Authenticate to Active Directory

by Bob McCormick

Does anyone know if there is a way to authenticate SER to MS Active Directory? I'd like to use an existing user database instead of having to manage users myself, and most of our users are in Active Directory. Has anyone tried this? What are most SER users using for authentication? Is anyone here tying into any kind of existing organization user database?

18 years, 11 months

[Serusers] ser as outbound proxy only

by jeff kwong

Hi to everybody! Im currently trying to setup an outbound proxy using ser. Im still a newbie with this ser and im not yet that good with the scripting. Could anyone tell me how to configure ser for outbound proxy only using rtpproxy? There are samples files in the Getting started manual but im having troubles eliminating the authentication/mysql part in script since im not yet familiar which modules depends on which. A striped down version of the nat-rtpproxy.cfg file would be really appreciated. Thanks in advance. _jeff

18 years, 12 months

[Serusers] ser+nat+siproxd+asterisk

by jeff kwong

Hi Guys! I just would like to share that I was able to get a working setup using SER as Softswitch, Asterisk as PSTN gateway and SIPROXD on my NAT Router. SIPROXD is an open source ALG and it effectively handles sip nat traversals. With it I dont have to run a seperate mediaproxy. When making calls from SIP UA to PSTN, RTP is as below: UA---NAT/SIPROXD---ASTERISK for 2 UA behind the same NAT: UA1--NAT---UA2 and for 2 UA behind different NATs: UA1--NAT1----NAT2---UA2 Thus there is less latency on signals and less traffic on SER. My question is, from the experience of other guys here, what do you think is the drawback or advantages of using SIPROXD together with SER to solve SIP NAT issues compared to other methods like using mediaproxy and rtpproxy?Will I still be able to do other SER features like accounting? Thanks! _jeff

19 years

[Serusers] Inserting SER into my voice network

by Ray Van Dolson

First off, here is my voice network layout currently: http://webdev.digitalpath.net/~rayvd/voice/voice_network2.png We're using Asterisk for voicemail, call routing (for long-distance, LNP, etc) and SER/rtpproxy at the other end which handles NAT onto private networks where customer's exist. This setup works fairly well for the most part, except that Asterisk does not have a jitter buffer. I would like to make use of rtpproxy (or mediaproxy) for their jitter buffer on both ends of our voice links here. To me, that would mean shoving a SER/rtpproxy combo between Asterisk and our provider network. Possibly on the same server. I could easily throw up a SER installation, but I'm trying to figure out if there's any way to leave Asterisk in the SIP path but remove it from the media path (have RTP just go straight from SER/rtpproxy to my provider's RTP proxy). Have any of you set up a scenario somewhat like this? Any recommendations? Asterisk CVS-HEAD does let you apply a patch and get some very alpha jitter buffer support. But CVS HEAD doesn't work reliably for me at all currently so I'm sticking with the latest stable release of 1.0.x series. I'm running SER 0.9.3 FYI on the SER proxies I have set up currently. Thanks. -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

19 years

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sr-users September 2005