sr-users March 2012

sr-users@lists.kamailio.org

87 participants
163 discussions

How to connect two legs of a call
by Chester Wood 15 Mar '12

15 Mar '12

Hello, SIP newbie here. We are a new WISP offering voip to our residential customers. Due to IP address scarcity we have to use large scale NAT in our network. In order for our local customers to call each other and have good call quality we wish to connect their ATA's directly to each other rather than have the RTP proxied 6000 miles across the public internet. I'm wondering about the best way to program an outbound proxy to monitor the calls and bypass the ITSP's B2BUA for local calls. We … [View More]want to rely on the ITSP's server for authentication (so we would only REGISTER our UA's locally after we get an OK from the remote server. Or perhaps we would not need a local registrar at all.) The remote server is also where the users manage their account. So we only want to attempt to connect the 2 UA's directly after the callee has sent an "OK" response to the INVITE from the ITSP. (The callee may have set "Do not disturb" on his account at the ITSP, and it needs to handle voicemail, forwarding etc.) We have other constraints. The server hardware needs to be very low power to be installed at our solar-powered head end. Thus for potentially several hundred clients it needs to handle only SIP, no media. We could potentially install a media relay server elsewhere, but hopefully we won't need to do any media proxying at all. The NAT assignment is all handled by a Mikrotik router at our head end. I've looked at the milkfish configuration file and it seems like in some ways, ours should be simpler. Milkfish assumes it is running on the gateway, so when it sets up an rtpproxy it doesn't cost anything. For us, it doesn't seem like we should have to do any NAT fixup or any media proxying. If the connection is local, the 2 UA's should connect directly with each other. Otherwise, the connection will be directly between our UA and whatever the ITSP sets up as the other end, and the ITSP already knows how to traverse our NAT very well. One idea I had is that, for local calls, when the initial INVITE is forwarded from the caller to the ITSP, the caller's original SDP payload is stashed away somewhere. Then when the INVITE comes through on the way to the callee, replace the ITSP's SDP info with the stashed original. Then when the callee responds with an "OK", I forward the response directly to the caller instead of the ITSP. There would have to be some fixup of loose ends, also. How to let the ITSP that its proxy services are no longer needed? (It could have decided to forward the call to voicemail in the meantime, etc.) Can you query the SER transaction database by Call-ID to find the other side of the call and extract information from it? Perhaps I'm barking up the wrong tree, but I'm sure there's a correct way to do this. Any help would be appreciated. thanks. [View Less]

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Re: [SR-Users] Enable session_id in ServerHello for TLS
by Daniel-Constantin Mierla 14 Mar '12

14 Mar '12

Hello, thanks for reporting back it's working -- please keep the mailing list cc-ed, so people looking for same issue will be able to find it when searching the web archive. I am using snom3xx with tls and kamailio 3.x a lot, never had issues, but I have no clue about the softphone.exe Cheers, Daniel On 3/11/12 8:09 PM, Kristijan Vrban wrote: > Hello Daniel, > > many thanks for the fast reply, And yes, the session_cache option > solved my problem. Well... the device i used … [View More]was the immemorial > snom360 softphone.exe > running with wine :) The softphone i use since years for TLS testing. > > Kristijan > > 2012/3/11 Daniel-Constantin Mierla<miconda(a)gmail.com>: >> Hello, >> >> >> On 3/11/12 1:28 AM, Kristijan Vrban wrote: >>> Hello, how to tell that Kamailio should juse a session_id for tls ? >>> See ssldump output below. I reckon that this is the reason the >>> client i use end with "handshake_failure". Because when is use >>> opensips, there is the session_id, and it's working. >>> >>> Kristijan >>> >>> 2 1 0.0228 (0.0228) C>S Handshake >>> ClientHello >>> Version 3.1 >>> cipher suites >>> TLS_RSA_WITH_RC4_128_MD5 >>> TLS_RSA_WITH_RC4_128_SHA >>> TLS_RSA_WITH_NULL_MD5 >>> TLS_RSA_WITH_NULL_SHA >>> TLS_DH_anon_WITH_3DES_EDE_CBC_SHA >>> TLS_DH_anon_WITH_RC4_128_MD5 >>> TLS_RSA_WITH_DES_CBC_SHA >>> TLS_RSA_EXPORT1024_WITH_RC4_56_SHA >>> TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA >>> TLS_DH_anon_WITH_DES_CBC_SHA >>> compression methods >>> NULL >>> 1 0.0519 (0.0519) C>S TCP FIN >>> 2 2 0.0432 (0.0204) S>C Handshake >>> ServerHello >>> Version 3.1 >>> session_id[0]= >>> >>> cipherSuite TLS_RSA_WITH_RC4_128_MD5 >>> compressionMethod NULL >>> 2 3 0.0432 (0.0000) S>C Handshake >>> Certificate >>> 2 4 0.0432 (0.0000) S>C Handshake >>> ServerHelloDone >>> 2 5 0.0452 (0.0020) C>S Alert >>> level fatal >>> value handshake_failure >>> 1 0.0744 (0.0225) S>C TCP FIN >>> 2 0.0681 (0.0228) S>C TCP FIN >> the tls module has now the option to turn on/off session caching, which was >> on by default in openser 1.x. Now it is off as it does not make much >> benefits with out multi-process architecture. Try to add to your config: >> >> modparam("tls", "session_cache", 1) >> >> Let me know if works -- the module parameter is missing from the readme, >> perhaps the author forgot to add it at the time of development -- I will try >> to sync the sources and the readme for tls module asap. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany >> http://www.asipto.com/index.php/kamailio-advanced-training/ >> -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/ [View Less]

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OT: is it allowed to send requests without waiting for the response of the previous request?
by Klaus Darilion 14 Mar '12

14 Mar '12

Hi all! I wonder if a presence server may send a NOTIFY if the previous NOTIFY in the dialog did not received an answer yet. I "grep"ed the relevant RFCs but couldn't find a definition. Thus, pointers are appreciated. Thanks Klaus

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kamailio issues??? or ekiga problem??
by Vineet Menon 14 Mar '12

14 Mar '12

Hi, I just installed kamailio for my sip testing. I am using it for simple registrar purposes... The config file is default file with debugging enabled. I registered 2 UAs with kamailio, but am not able to call using aor dialing. I inspected packets with wireshark, which i think is a bit problematic. my server's at 31.21.33.238, while UAs are at 31.21.33.250 and 31.21.33.236. I have attached the wireshark file for packets. see packet number.5 and 7. Why has ekiga put the from field as ip:0.… [View More]

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rfc: unique id for profiles
by Daniel-Constantin Mierla 13 Mar '12

13 Mar '12

Hello, from the last IRC devel meetings discussions, one topic was about adding an unique id field per user profiles, with the decision to bring it to mailing lists for broader audience. That will help to reference easier across records. At this moment, the most relationships are via (username) or (username,domain) when use_domain=1, bringing some complexity. The main reasons I came up to this were: - in the perspective of gruu implementation for usrloc, with global addresses, an unique … [View More]

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Bad AVP Flags When Using avp_db_load()
by Simpson Chua 13 Mar '12

13 Mar '12

Hi Daniel, Thanks! I changed to use the $avp() syntax and that fixed it. The examples I followed was from an old wiki. Simpson ________________________________ From: "sr-users-request(a)lists.sip-router.org" <sr-users-request(a)lists.sip-router.org> To: sr-users(a)lists.sip-router.org Sent: Tuesday, March 13, 2012 12:45 PM Subject: sr-users Digest, Vol 82, Issue 41 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or … [View More]

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Compilation errors...
by Krishna Kurapati 13 Mar '12

13 Mar '12

Hi, git cloned main branch and tried to compile it on Cent OS 6. The mi_xmlrpc module is failing to compile make Makefile:46: You are using an unsupported libxmlrpc-c3 version (1.16.24), compile at your own risk! Makefile:46: You are using an unsupported libxmlrpc-c3 version (1.16.24), compile at your own risk! CC (gcc) [M mi_xmlrpc.so] abyss_conf.o CC (gcc) [M mi_xmlrpc.so] abyss_conn.o CC (gcc) [M mi_xmlrpc.so] abyss_data.o abyss_data.c:219: error: expected ‘)’ before ‘*’ token … [View More]

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Bad AVP Flags When Using avp_db_load()
by Simpson Chua 13 Mar '12

13 Mar '12

Hi, I'm trying to use avp_db_load to check for some IPs but I'm getting bad avp flags errors. As far as I understand, the syntax is correct. TableName: labservers uuid username domain attribute value type "" "" "domain1" "lab1" "192.168.55.1" 0 "" &… [View More]

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log not visible in stderr
by Vineet Menon 13 Mar '12

13 Mar '12

Hi, I am new to kamailio and want to install a simple registrar. Now, I have a ubuntu machine in which I have installed kamailio 3.2.x. I have included #!define WITH_DEBUG in configuration file, while the rest of the file is same as default. But I am not able to see any debug message. Any help? Regards, Vineet Menon

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exec_avp problem
by Uri Shacked 12 Mar '12

12 Mar '12

Hi, i am using the exec_avp command as followed: *exec_avp("[/usr/local/etc/kamailio/maintenance/service/service_pr.sh", "$avp(xxx),$avp(yyy)");* ** here is the shell script content: ** *RETVAL=-1 RES=`kamctl cr reload` RETVAL=$? echo "$RETVAL" echo "$RES"* ** I keep on getting 127 as the retval. any ideas why? I must say that two days ago, the script was OK..... :-( where i run the script manualy i have no problems... retval is 0. here is the log with debug 4: *cfg_set_now(): core.debug … [View More]

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sr-users March 2012