Re: [SR-Users] Kamailio -PSTN Gateway
Morning Daniel:
Thank you very much for your response!
As for the routing explanation, I am afraid, I couldn't narrate the issue. I
rephrase it, here below:
1: By default Kamailio listens on all interfaces (implying that it has
knowledge of all interfaces and corresponding subnets, please correct me if
wrong.
2: A packet arriving at say 10.10.10.1/8 will be processed by the corresponding
module
3: A packet destined for 192.168..1.15/24 (say our PSTN GW/ MGC) how will it be
routed? Wil it be sent on all interfaces wlan0 (above) and eth0 with same
subnet as eth0 or only sent out via eth0?
Thanks again!
KR,
Zaka
From: sr-users-boun...@lists.sip-router.org
[sr-users-boun...@lists.sip-router.org] on behalf of Daniel-Constantin Mierla
[mico...@gmail.com]
Sent: Monday, September 29, 2014 9:36 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio -PSTN Gateway
Hello,
On 24/09/14 08:44, Zaka Ul Isam wrote:
Hello Folks:
Please help with above, I have browsed and tried various suggestions on this
list without much luck!
I think problem can be reduced to three questions ;)
1: Apart from DEFINE WITH PSTN directive, do I need certain modules to be
compiled? (DIALPLAN, CARRIERROUTE, LCR aso) If yes, then we ought to put up
a list of modules required for each task/ activity.
if you have only a pstn gateway, it is not necessary to add extra modules.
If you have more, makes sense to get lcr or an alternative, the right
place with be to replace some of the content in route PSTN with
functions from the new module.
2: How Kamailio core decides the call routing? i.e: based on dialed digits as
specified in dialplan .
# - update the condition to match your dialing rules for PSTN routing
if(!($rU=~^(\+|00355)[0][4-9]{3,20}$))
return;
Yes, in the case of default config file, but you can change that as
needed by you. Default config file was made mainly to give an example of
routing to PSTN, but can be update to match your rules for sending to PSTN.
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = 10.0.0.101 desc My PSTN GW Address
#
# - by default is empty to avoid misrouting
pstn.gw_ip = 10.1.6 desc PSTN GW Address
pstn.gw_port = 5080 desc PSTN GW Port
#!endif
3: If Kamailio is listening on say wlan0 interface 10.42.0.1:5060 (Ubuntu HOT
SPOT) can it route to PSTN Gateway/ Softswitch on P2P1 (Eth0) without
Bridging? Put another way, does kernel route towards the specified interface
or Kamailio is capable of routing based on active routing cache?
This is a matter of your routing rules in kernel. Also, if you do
natting there, the PSTN gateway must be able to do nat traversal for rtp.
Cheers,
Daniel
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[SR-Users] Kamailio with Jitsi - Presence trouble
Hello all!I tried sending this earlier too but since I was not subscribed to the mailing list, it bounced back.I am trying to setup a secure videoconferencing setup formy small office. After a lot of convincing, my bosses have allowed meto create a setup and have given me a live IP. I have used it on aUbuntu 12.04 setup and want to setup a SIP server for very few users(less than 10, at most) to use it through Jitsi. I am trying to followthe tutorials available at:http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehourhttps://www.johncahill.net/wiki/index.php/SkypelikeconferencingSystemIam able to log into Jitsi on independent machines, both behind andwithout a firewall. But the trouble I am facing is that the users do notappear online to each other. If I try to send a text messagenonetheless, I get the 403 Not allowed error.At first, I triedthe config file for Kamailio provided on the first link. It had someproblems due to it being an old version and the config fil e on the 2ndlink helped me out. I suspect there is some problem with the PRESENCEmodule but I do not know what since it does not give any errorwhatsoever. I added the #!define WITHPRESENCE line and installed thePresence module but to no avail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00b for Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a week now. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] R: Re: R: Re: RTPPROXY BRANCH
Unfortunately rtpengine doesn't work in this way. At the end of the calls this is the output log: Final packet stats: Tag 'Fw3D7R0', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30224192.168.10.20:7078 , 540 p, 92880 b, 0 e Media #1, port 30225192.168.10.20:7079 (RTCP), 3 p, 324 b, 0 e Media #2, port 30256192.168.10.20:9078 , 0 p, 0 b, 0 e Media #2, port 30257192.168.10.20:9079 (RTCP), 3 p, 264 b, 0 e Tag 'qWE6Gsh', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30140192.168.10.50:7078 , 533 p, 91068 b, 0 e Media #1, port 30141192.168.10.50:7079 (RTCP), 5 p, 444 b, 0 e Media #2, port 30170192.168.10.50:9078 , 0 p, 0 b, 0 e Media #2, port 30171192.168.10.50:9079 (RTCP), 1 p, 88 b, 0 e Tag 'TTPyT~Hdw', created 0:41 ago, in dialogue with 'Fw3D7R0' Media #1, port 30206172.20.11.208:7078 , 1070 p, 183736 b, 0 e Media #1, port 30207172.20.11.208:7079 (RTCP), 4 p, 496 b, 0 e Media #2, port 30240172.20.11.208:9078 , 4188 p, 1435946 b, 0 e Media #2, port 30241172.20.11.208:9079 (RTCP), 4 p, 400 b, 0 e 192.168.10.x clients are natted..it seems that rtpengine open 2 ports (for example video) for each receiver (30256 amp; 30170) and 1 port for the caller (30240). But on the INVITE of Kamailio only video port 30170 is offered to receivers, instead on caller side there are 2 distinct 183s message that offer 30190 amp; 30240. It's a little bit strange because some of these port doesn't appear in the log of rtpengine. At the end I can see video only on one receiver I don't know if the problem is on Kamailio (rtpproxy-ng module) or in the rtpengine :) Without rtpproxy: - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted client so no needs of rtpproxy) - B offers port b1,b2 (183) - C offers port c1,c2 (182). - A starts to send audio/video RTP to B on port b1,b2 - A starts to send audio/video RTP to C on port c1,c2 With rtpproxy: - A offers port a1,a2 (audio video) in INVITE to Kamailio - Kamailio contact rtpproxy because Bamp;C are natted clients - rtpproxy check callid and offer offers port k1,k2 - Kamailio sends INVITE to B offering k1,k2 - Kamailio sends INVITE to C offering k1,k2 - B offers port b1,b2 (183) - C offers port c1,c2 (182) - Kamailio sends 183 to A (for B leg) offering p1,p2 - Kamailio sends 183 to A (for B leg) offering p3,p4 - A starts to stream on p1,p2,p3,p4 but only one receiver can see the video (B or C depends who will be the first:)) I don't know if it depends on that B amp; C receives same ports; i don't know if rtpproxy is able to duplicate stream received from A to all receiver If A sends two streams, there is no need for duplication. A sending to p1 should be forwarded to B (b1) and A sending to p3 should be forwarded to C (c1). Both should be able to receive media sent by A. I believe that's what rtpengine does (but I haven't tested it). The reverse direction might be more confusing, but a final 200 OK with SDP should fix that. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Uri transformations stopped working after upgrade from 3.3 to 4.1.6.
Hello,
Some Uri transformations stopped working after upgrade from 3.3 to 4.1.6.
*To: sip:01412127775678;cic=012;csel=noind@10.25.153.150:5060
http://noind@10.25.153.150:5060;user=phone*
*In v3.3*
$tu({uri.params}) return: cic=012;csel=noind
$tu({uri.param,cic}) return:012
*In v4.1.6*
$tu({uri.params}) return : user=phone
$tu({uri.param,cic}) return :
Best regards,
Julia
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Re: [SR-Users] Questions about multiple contacts in kamailio
Hi, you can achieve this by using the flag 0x04 when calling save() function from registrar module: http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1965144 Regards, Federico On 30 Sep 2014 10:54, qw applema...@163.com wrote: Hi, I have one question about contact in kamailio. Sometimes, one user may register multiple contacts in kamailio server, where contacts are represented as ip:port. For example, the user registers at first, and lose connection to the internet later. After one minute, the user connects the internet again, but can't get service from kamailio server. Then the user need to registers with new contact. Now there are two contacts in kamailio, i.e. old one and new one. How can I remove the old contact, and make sure there is only one contact in kamailio server. Looking forward to your help! B.R. andrew ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Uri transformations stopped working after upgrade from 3.3 to 4.1.6.
$(tu{uri.uparam,cic}) gets ERROR wrong format,
$(tu{uri.uparam}) return : phone
So, uparam return the value of parameter user .
I solved a problem by usage $(tu{param.value,cic,;})
BR
Julia
On Tue, Sep 30, 2014 at 12:09 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
On 30/09/14 10:41, Julia Boudniatsky wrote:
Hello,
Some Uri transformations stopped working after upgrade from 3.3 to 4.1.6.
*To: sip:01412127775678;cic=012;csel=noind@10.25.153.150:5060
http://noind@10.25.153.150:5060;user=phone*
*In v3.3*
$tu({uri.params}) return: cic=012;csel=noind
$tu({uri.param,cic}) return:012
*In v4.1.6*
$tu({uri.params}) return : user=phone
$tu({uri.param,cic}) return :
looks like this is happening as a fix, as the cic is an username
parameter, not a uri parameter.
You should use {uri.uparam,cic}:
http://www.kamailio.org/wiki/cookbooks/devel/transformations#uriuparam
Cheers,
Daniel
--
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http://www.linkedin.com/in/miconda
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Re: [SR-Users] Uri transformations stopped working after upgrade from 3.3 to 4.1.6.
On 30/09/14 11:46, Julia Boudniatsky wrote:
$(tu{uri.uparam,cic}) gets ERROR wrong format,
$(tu{uri.uparam}) return : phone
So, uparam return the value of parameter user .
Right, I forgot what I implemented and misread the docs.
You should add a transformation to return the user parameters.
I solved a problem by usage $(tu{param.value,cic,;})
Be careful not to have cic as last parameter before @.
Might be safer to use:
$(tu{s.select,0,@}{param.value,cic})
Cheers,
Daniel
BR
Julia
On Tue, Sep 30, 2014 at 12:09 PM, Daniel-Constantin Mierla
mico...@gmail.com mailto:mico...@gmail.com wrote:
Hello,
On 30/09/14 10:41, Julia Boudniatsky wrote:
Hello,
Some Uri transformations stopped working after upgrade from 3.3
to 4.1.6.
*To: sip:01412127775678;cic=012;csel=noind@10.25.153.150:5060
http://noind@10.25.153.150:5060;user=phone*
_In v3.3_
$tu({uri.params}) return:cic=012;csel=noind
$tu({uri.param,cic}) return: 012
_In v4.1.6_
$tu({uri.params}) return : user=phone
$tu({uri.param,cic}) return :
looks like this is happening as a fix, as the cic is an username
parameter, not a uri parameter.
You should use {uri.uparam,cic}:
http://www.kamailio.org/wiki/cookbooks/devel/transformations#uriuparam
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda http://twitter.com/#%21/miconda
-http://www.linkedin.com/in/miconda
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Re: [SR-Users] Kamailio with Jitsi - Presence trouble
Hello, you should grab the SIP traffic on kamailio server and send it to the mailing list. In this way we can spot why you get the 403 Not Allowed. You can use: ngrep -d any -qt -W byline sip port 5060 Cheers, Daniel On 30/09/14 08:50, Gaurav Kumar wrote: Hello all! I tried sending this earlier too but since I was not subscribed to the mailing list, it bounced back. I am trying to setup a secure video-conferencing setup for my small office. After a lot of convincing, my bosses have allowed me to create a setup and have given me a live IP. I have used it on a Ubuntu 12.04 setup and want to setup a SIP server for very few users (less than 10, at most) to use it through Jitsi. I am trying to follow the tutorials available at: http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour https://www.johncahill.net/wiki/index.php/Skype_like_conferencing_System I am able to log into Jitsi on independent machines, both behind and without a firewall. But the trouble I am facing is that the users do not appear online to each other. If I try to send a text message nonetheless, I get the 403 Not allowed error. At first, I tried the config file for Kamailio provided on the first link. It had some problems due to it being an old version and the config file on the 2nd link helped me out. I suspect there is some problem with the PRESENCE module but I do not know what since it does not give any error whatsoever. I added the #!define WITH_PRESENCE line and installed the Presence module but to no avail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL (I have hidden my live IP in the text). My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8e I can upload the logs/other file content as needed. I am a n00b for Kamailio so you will have to be patient with me. Please help me out here. I have been after it for almost a week now. Cheers! Gaurav Dear *sr-users!* Get Yourself a cool, short *@in.com* Email ID now! http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Call Group versus MAX_BRANCHES limit
Hello, On 29/09/14 22:47, João Vitor Arruda wrote: Hi folks, I have a question related with the limited number of branches being 12 in config.h #define MAX_BRANCHES12 /*! maximum number of branches per transaction */ I am trying to implement a Call Group that consists in trying each member of the group (that can result in a parallel forking when lookup() is used) in sequence (similar to serial forking). Currently I have written code that is similar to the one described here: http://www.kamailio.org/dokuwiki/dokuphp/tutorials:avpops#serial_forking The only difference is that for each member I call lookup() and then in the failure_route(), I pull the next member from the stack and call lookup() again. The code works fine when used for groups with less than 12 members. However, call groups with 12 or more members fails as I can never reach remaining extensions once MAX_BRANCHES limit is reached. I've tried another approach using the functions described here: http://kamailio.org/docs/modules/4.1.x/modules/tm.html#tm.serial_forking but it too uses a new branch for every group member until the 12 limit is reached out. I also tried to use the functions remove_branch(index) and clear_branches() (both of which are poorly documented. In fact the only reference I ever found for these functions was here: http://www.kamailio.org/wiki/features/new-in-3.2.x#functions). Unfortunately, I wasn't able to prevent the MAX_BRANCHES limit from being reached. Ultimately, my goal is to have a limitless Call Group. Do you have any suggestions? (other than increasing the hard coded MAX_BRANCHES limit) the easiest is probably recompiling with more branches. There was someone saying that he is going to submit a patch on making the number of branches more dynamic, but I haven't seen it back. From routing point of view, you can try a workaround with: - append 11 branches to the same sip address (see append_branch() function) and relay - be sure you allow traffic from server itself - now you get 12 INVITE coming back to kamailio, so you get 12 INVITE requests and you can set 12 different destinations for each, ending up with 144 over-all branches in the group call - if you need more, you can loop back again one or more of those INVITE requests with branches pointing to same SIP address Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] dlg_set_timeout_by_profile not working on kamailio 4.1.5
I'm trying to use dlg_set_timeout_by_profile and it doesn't do nothing on Kamailio 4.1.5 If i try to use the dlg_set_timeout, it also doesn't work, but it gives the following error CRITICAL: dialog [dlg_timer.c:205]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f9825ac6880 tl-next=(nil) tl-prev=(nil) ERROR: dialog [dlg_hash.c:1094]: update_dlg_timeout(): failed to update dialog lifetime If i set the timeout with the $avp, it only works before dlg_manage(), but also with a Warning WARNING: dialog [dlg_handlers.c:1245]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7f9da0dc7f08 [4066:9857] with clid 'bba73ace3ee94b95bf5b1782406047bd' and tags '6460df6b8b3947ba96b1cf65330bf524' 'm2pmrUpHN06Fc' The dialog module configurations are: modparam(dialog, db_url, DBURL) modparam(dialog, dlg_flag, FLT_DLG) modparam(dialog, db_mode, 0) modparam(dialog, enable_stats, 1) modparam(dialog, dlg_match_mode, 1) modparam(dialog, profiles_with_value, user ; domain ; permissions) modparam(dialog, profiles_no_value, inbound ; outbound ; all) modparam(dialog, send_bye, 1) modparam(dialog, default_timeout, 21600) modparam(dialog, ka_timer, 0) modparam(dialog, ka_interval, 0) modparam(dialog, timeout_avp, $avp(dlgtimeout)) I'm very interested to use dlg_set_timeout_by_profile, any idea about what is wrong with that ? Regards, *José Ferreira | Technical Manager* M. +351 91 775 7166 | jferre...@wavecom.pt Wavecom, Soluções Rádio, SA Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 234 919 190 | F. +351 234 919 191 *www.wavecom.pt http://www.wavecom.pt/* [image: WavecomSignature] ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio with Jitsi - Presence trouble
Hi,I ran the code on my server (as Daniel suggested) and then tried to send message from one user to other but no traffic was detected.ngrep d any qt W byline sip port 5060interface: anyfilter: (ip or ip6) and ( port 5060 )match: sipAlso, after I tried to install the PRESENCE module and having restarted the systems today, I noticed that the response that I am getting on Jitsi is 403 Not Relaying and not 403 Not Allowed. I am not sure what caused the change.I hope someone can help me figure it out.Cheers!Gaurav Original message From:DanielConstantin Mierla mico...@gmail.com Date: 30 Sep 14 16:19:30Subject: Re: [SRUsers] Kamailio with Jitsi Presence troubleTo: ; Kamailio (SER) Users Mailing ListHello,you should grab the SIP traffic on kamailio server and send it to the mailing list. In this way we can spot why you get the 403 Not Allowed. You can use:ngrep d any qt W byline sip port 5060Cheers, DanielOn 30/09/14 08:50, Gaurav Kumar wrote:Hello all!I tried sendin g this earlier too but since I was not subscribed tothe mailing list, it bounced back.I am trying to setup a secure videoconferencing setup for mysmall office. After a lot of convincing, my bosses have allowed meto create a setup and have given me a live IP. I have used it on aUbuntu 12.04 setup and want to setup a SIP server for very fewusers (less than 10, at most) to use it through Jitsi. I am tryingto follow the tutorials available at: http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehourhttps://www.johncahill.net/wiki/index.php/SkypelikeconferencingSystemI am able to log into Jitsi on independent machines, both behindand without a firewall. But the trouble I am facing is that theusers do not appear online to each other. If I try to send a textmessage nonetheless, I get the 403 Not allowed error.At first, I tried the config file for Kamailio provided on thefirst link. It had some problems due to it being an old versionand the config file on the 2nd link help ed me out. I suspect thereis some problem with the PRESENCE module but I do not know whatsince it does not give any error whatsoever. I added the #!defineWITHPRESENCE line and installed the Presence module but to noavail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00bfor Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a weeknow. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now! SIP Express Router (SER) and Kamailio (OpenSER) srusers mailing list srus...@lists.siprouter.org http://lists.siprouter.org/cgibin/mailman/listinfo/srusersDanielConstantin Mierla http://twitter.com/#!/miconda http://www.linkedin.com/in/micondaGet Yourself a cool, short @in.com Email ID now! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Call Group versus MAX_BRANCHES limit
Hi Daniel, would it be inappropriate to increase the branch number to the maximum possible, by default? Regards, Carlos On Tue, Sep 30, 2014 at 5:28 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 29/09/14 22:47, João Vitor Arruda wrote: Hi folks, I have a question related with the limited number of branches being 12 in config.h #define MAX_BRANCHES12 /*! maximum number of branches per transaction */ I am trying to implement a Call Group that consists in trying each member of the group (that can result in a parallel forking when lookup() is used) in sequence (similar to serial forking). Currently I have written code that is similar to the one described here: http://www.kamailio.org/dokuwiki/dokuphp/tutorials:avpops#serial_forking The only difference is that for each member I call lookup() and then in the failure_route(), I pull the next member from the stack and call lookup() again. The code works fine when used for groups with less than 12 members. However, call groups with 12 or more members fails as I can never reach remaining extensions once MAX_BRANCHES limit is reached. I've tried another approach using the functions described here: http://kamailio.org/docs/modules/4.1.x/modules/tm.html#tm.serial_forking but it too uses a new branch for every group member until the 12 limit is reached out. I also tried to use the functions remove_branch(index) and clear_branches() (both of which are poorly documented. In fact the only reference I ever found for these functions was here: http://www.kamailio.org/wiki/features/new-in-3.2.x#functions). Unfortunately, I wasn't able to prevent the MAX_BRANCHES limit from being reached. Ultimately, my goal is to have a limitless Call Group. Do you have any suggestions? (other than increasing the hard coded MAX_BRANCHES limit) the easiest is probably recompiling with more branches. There was someone saying that he is going to submit a patch on making the number of branches more dynamic, but I haven't seen it back. From routing point of view, you can try a workaround with: - append 11 branches to the same sip address (see append_branch() function) and relay - be sure you allow traffic from server itself - now you get 12 INVITE coming back to kamailio, so you get 12 INVITE requests and you can set 12 different destinations for each, ending up with 144 over-all branches in the group call - if you need more, you can loop back again one or more of those INVITE requests with branches pointing to same SIP address Cheers, Daniel -- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] R: Re: R: Re: RTPPROXY BRANCH
Hi Richard, this is more or less the same problem that I am experiencing. To understand it better, just assume one branch needs to do SRTP, and the other simple RTP. To make this happen, you will have to enable rtpengine differently for the same call, and this is where the crash/error happens. Regards, Carlos On Tue, Sep 30, 2014 at 2:51 AM, Marino Mileti marino.mil...@alice.it wrote: Unfortunately rtpengine doesn't work in this way. At the end of the calls this is the output log: Final packet stats: Tag 'Fw3D7R0', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30224192.168.10.20:7078 , 540 p, 92880 b, 0 e Media #1, port 30225192.168.10.20:7079 (RTCP), 3 p, 324 b, 0 e Media #2, port 30256192.168.10.20:9078 , 0 p, 0 b, 0 e Media #2, port 30257192.168.10.20:9079 (RTCP), 3 p, 264 b, 0 e Tag 'qWE6Gsh', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30140192.168.10.50:7078 , 533 p, 91068 b, 0 e Media #1, port 30141192.168.10.50:7079 (RTCP), 5 p, 444 b, 0 e Media #2, port 30170192.168.10.50:9078 , 0 p, 0 b, 0 e Media #2, port 30171192.168.10.50:9079 (RTCP), 1 p, 88 b, 0 e Tag 'TTPyT~Hdw', created 0:41 ago, in dialogue with 'Fw3D7R0' Media #1, port 30206172.20.11.208:7078 , 1070 p, 183736 b, 0 e Media #1, port 30207172.20.11.208:7079 (RTCP), 4 p, 496 b, 0 e Media #2, port 30240172.20.11.208:9078 , 4188 p, 1435946 b, 0 e Media #2, port 30241172.20.11.208:9079 (RTCP), 4 p, 400 b, 0 e 192.168.10.x clients are natted..it seems that rtpengine open 2 ports (for example video) for each receiver (30256 30170) and 1 port for the caller (30240). But on the INVITE of Kamailio only video port 30170 is offered to receivers, instead on caller side there are 2 distinct 183s message that offer 30190 30240. It's a little bit strange because some of these port doesn't appear in the log of rtpengine. At the end I can see video only on one receiver I don't know if the problem is on Kamailio (rtpproxy-ng module) or in the rtpengine :) Without rtpproxy: - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted client so no needs of rtpproxy) - B offers port b1,b2 (183) - C offers port c1,c2 (182). - A starts to send audio/video RTP to B on port b1,b2 - A starts to send audio/video RTP to C on port c1,c2 With rtpproxy: - A offers port a1,a2 (audio video) in INVITE to Kamailio - Kamailio contact rtpproxy because BC are natted clients - rtpproxy check callid and offer offers port k1,k2 - Kamailio sends INVITE to B offering k1,k2 - Kamailio sends INVITE to C offering k1,k2 - B offers port b1,b2 (183) - C offers port c1,c2 (182) - Kamailio sends 183 to A (for B leg) offering p1,p2 - Kamailio sends 183 to A (for B leg) offering p3,p4 - A starts to stream on p1,p2,p3,p4 but only one receiver can see the video (B or C depends who will be the first:)) I don't know if it depends on that B C receives same ports; i don't know if rtpproxy is able to duplicate stream received from A to all receiver If A sends two streams, there is no need for duplication. A sending to p1 should be forwarded to B (b1) and A sending to p3 should be forwarded to C (c1). Both should be able to receive media sent by A. I believe that's what rtpengine does (but I haven't tested it). The reverse direction might be more confusing, but a final 200 OK with SDP should fix that. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] rtpengine alt stream question
Hello, I recently encountered SDP attributes that look like this: Media Attribute (a): alt:1 3 : njfxofkg bavpjfxg 192.168.2.59 41978 Media Attribute (a): alt:2 2 : fpukfyaj dqmiarjx 192.168.111.1 41978 Media Attribute (a): alt:3 1 : euvrwenk jctmhavh 192.168.238.1 41978 I was not familiar with these. However, RFC 4796 says: alt: the media stream is taken from the alternative source. A typical use case for this is an event where the ambient sound is separated from the main sound. The alternative audio stream could be, for example, the sound of a jungle. Another example is the video of a conference room, while the main stream carries the video of the speaker. This is similar to the 'live' role in H.239. RFC 6064 http://tools.ietf.org/html/rfc6064#section-4.4 speaks to this a little as well. This does not really make any clearer to me what these mean in this scenario, and unfortunately I cannot find out from the user. My question is: I assume that rtpengine (both Kamailio and daemon-side) does not support this, correct? Are there any plans to in the future? Can anyone with more experience with video, perhaps, give some insight into the meaning and application of this attribute? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] simplest config file possible - not working!
Sorry Guys, I'm very new to this. Taking a dive 'under the hood', to try to
learn what is going on, and to see what I can achieve.
I have 1 sip device ( voip telephone) on the network, constantly firing out
REGISTER requests (I have turned off my asterisk server!). That's good.
I have a terminal window open, running tcpdump. That's good too - I can see
what is going on.
I also have kamailio - giving me absolutely nothing. It's not complicated
config file. In fact it is the simplest I could do. However, it logs
nothing - and i am expecting it to.
Can someone put me out of my misery, and point out the glaring error I have?
How can I send messages to STDERR while testing?
Cheers,
Phil
## WARNING - FOR TESTING EDUCTION ONLY
### Global Parameters #
debug=5
fork=no
log_stderror=yes
port=5060
### Modules Section
# set module path
mpath=/usr/local/lib/kamailio/modules/
loadmodule sl.so
### Routing Logic
route{
if(method==REGISTER) {
log(1,this SIP request is a REGISTER message\n);
sl_send_reply(404, No registrar);
};
if (af!=INET6) {
log(1,Message received over IPv4 link\n);
};
if(proto==UDP) {
log(1,SIP message received over UDP\n);
};
if(status==200) {
log(1,this is a 200 OK reply\n);
};
if(uri==myself) {
log(1,the request is for local processing\n);
};
}
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