sr-users September 2014

sr-users@lists.kamailio.org

96 participants
171 discussions

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:23, Marino Mileti wrote: > The problem isn't on 183s but on the multiple INVITE that Kamailio sends to > clients behind rtpengine. Rtpengine open new ports for answer but on INVITE > the rtpengine ports are the same...This happens because for all these > clients the callid is still the same..so for rtpengine there's no > difference...for this reason I can see early media only on one of the > receivers I still don't understand what difference an RTP proxy is supposed to make for the invite. Your offering client only opens one endpoint and sends that out. Without RTP proxy, all receiving clients see the same endpoint and they will all send their early media to that same endpoint. With RTP proxy, the same thing happens. What's the difference? cheers

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Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:29, Marino Mileti wrote: > Wow! Do you have an example of how to do that? How I have to modify my > kamailio.conf in order to instructs rtpproxy to user from-tag & to-tag in > this way? You don't have to do anything, tags are already included in all the messages. cheers

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Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:08, Marino Mileti wrote: > But with from-tag and To-tag it's possible to instruct rtpengine to generate > new couple of ports for each branch of a call? In the source code of > rtpengine it seems that it check only the callid parameter Yes it will. The call-id is only a vague umbrella under which the from/to tags are collected. All media parameters, ports etc are associated with the tags, and a single call can contain any number of different from/to tags, all with their own separate parameters. cheers

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R: Re: R: Re: RTPPROXY & BRANCH
by Marino Mileti 29 Sep '14

29 Sep '14

No no. The video will be sent by the caller user to all the callees. I'l try to explain better. My scenario is: - A make a call to a group... B & C are group member...so Kamailio is able to call them in parallel using alias.. - B & C receive the INVITEs & replies with two separate 183 with SDP (in SDP they specified where they are able to receive audio&video) - A receives two 183...& starts to send its RTP video stream to B & C (early media) - Once B or C answers the call the other leg is cancelled.. Everything is working fine but if B & C are behind NAT rtpproxy is activated and during INVITE for B &C rtprpoxy doesn't generate a couple of new ports for each of them but it uses always the same ports. So the only fastest client (B or C) get the video. I don't want to change IP between 183 & 200, i would like only that rtpptoxy sends INVITE for B & C with differents port.Is it possible to implement this scenario or there's some turnarund? ----Messaggio originale---- Da: frank(a)carmickle.com Data: 25-set-2014 16.22 A: "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] R: Re: RTPPROXY & BRANCH On Sep 25, 2014, at 10:09 AM, Marino Mileti <marino.mileti(a)alice.it> wrote:Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) I don't understand. Are you saying that you have clients that when they receive an invite sent video with 183? How do you want to composite the video to show to the caller? It is not RFC3261 compliant to change IP and port from 183 to 200. Of course you can reinvite after the 200. Most B2BUAs require you to ignore early media and generate something locally to send to the caller or just send them 180. Maybe if you explain your use case someone can help you. --FC

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Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/26/14 16:57, Marino Mileti wrote: > Hello, > >> On Friday 26 September 2014 16:44:44 Marino Mileti wrote: >>> Hi guys, >>> I've seen that setting the parameter extra_id_pv, every branch should >>> be a different callid.. >>> How can i set this parameter? I've tried with : >>> modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)") >>> >>> but in the INVITE message the callid is still the same for all >>> branches. Any suggest? > >> Did you assign a value to $avp(extra) in the script, before calling any of > the rtpproxy functions? > >> Did you use the 'b' parameter in the call to rtpproxy_*() functions? >> -- >> Alex Hermann > > Yes, i've assigned a value in the script, just to try I've assigned $rU and > on the log of rtpengine I can see the value of "via-branch" valorized > correctly. I've also used the 'd' parameter in rtpproxy_manage() function. > > I've also checked on the source code of rtpengine and the part regarding the > "parse & get" of via-branch value is commented...I'm using the master branch > of rtpengine. > > This is the piece of code of call_interfaces.c where the via-branch > parameter is commented: > //bencode_dictionary_get_str(input, "via-branch", &viabranch); The via-branch is currently unused in rtpengine, even though it does appear in the control messages. From-tag and to-tag are sufficient to identify a unique dialogue within a call. cheers

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R: Re: RTPPROXY & BRANCH
by Marino Mileti 29 Sep '14

29 Sep '14

Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) So at the moment this scenario is not possible? Is there any possible turnaround? ----Messaggio originale---- Da: rfuchs(a)sipwise.com Data: 24-set-2014 1.15 A: <sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] RTPPROXY &amp; BRANCH On 23/09/14 04:35 AM, marino.mileti(a)alice.it wrote: > I've a problem with rtpproxy during a parallel ring scenario. > I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I > try to call them in parallel mode (ringall) rtpproxy module sends in to > the INVITE the same RTP ports. > > Is it possible to manage rtpproxy in order to generate a couple of new > port for each branch? Not at the moment. The logic is that one endpoint advertised by the client is mapped to one endpoint advertised by rtpengine. Why would you like a separate set of ports? cheers _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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rtpproxy_offer and fix_nated_sdp in one route
by Sebastian Damm 29 Sep '14

29 Sep '14

Hi, I switched from rtpproxy module to the rtpproxy-ng module lately, and noticed a strange behavior. In my branch route to the device, I have two statements: fix_nated_sdp("1"); rtpproxy_offer(); The first command appends a line with "direction:active" to the SDP. The second one puts the RTP proxy in the stream. This worked all well with the old rtpproxy module. But with the new rtpproxy-ng module, I get an empty line after the SDP body, just before the "direction:active" line in the SDP, which makes the packet invalid. I tried changing the order of both statements, but without any difference. What I saw is, that the old module just sent some basic parameters to the rtpproxy and got only IP and port back. The new module sends the complete SDP to the rtpengine and gets back the "fixed" SDP. Has anyone ever seen this? Is there a way to fix it? Best Regards, Sebastian

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event_route[core:receive-parse-error] is not executed 4.1.6
by Julia Boudniatsky 29 Sep '14

29 Sep '14

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Event-parse-error: $rm from $avp(inc_carrier)/n$mb/n"); } corelog=1 debug=0 kamailio 4.1.6 from source Wrong header "From" is simulated by SIPP. In log I receive only ERROR:<core> [parser/parse_from.c:79]: parse_from_header() ...... Thank you, Julia

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Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 29 Sep '14

29 Sep '14

Hello all!I am trying to setup a secure videoconferencing 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:skypelikeserviceinlessthanonehour&https://… 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 WITHPRESENCE" 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/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!

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event_route[core:receive-parse-error] is not executed 4.1.6
by Julia Boudniatsky 29 Sep '14

29 Sep '14

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Parse-error: $rm from $avp(inc_carrier)/n$mb/n"); }

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sr-users September 2014