Re: [SR-Users] RTPengine + kernel module long call RTP drops

On 05/04/17 02:53 PM, Anthony Joseph Messina wrote: On Wednesday, April 5, 2017 8:55:36 AM CDT Richard Fuchs wrote: On 04/04/2017 09:33 PM, Anthony Joseph Messina wrote: After more digging, I see (from the Asterisk perspective) that after a certain amount of time, the "RTCP report"

Re: [SR-Users] RTPengine + kernel module long call RTP drops

On 04/04/2017 09:33 PM, Anthony Joseph Messina wrote: After more digging, I see (from the Asterisk perspective) that after a certain amount of time, the "RTCP report" size gets smaller and this is the point at which the audio from Asterisk back to the softphone is dropped. Again, this audio dro

Re: [SR-Users] rtpengine key file

On 02/09/2017 05:05 AM, Kjeld Flarup wrote: OK The recipees I've found for rtpengne.conf, does not match /etc/rtpengine/rtpengine.sample.conf Primarily the error is caused by not having [rtpengine] in /etc/rtpengine/rtpengine.conf Looks like some change of file format. Support for a rea

Re: [SR-Users] RTPEngine disable/enable crash kamailio

On 02/06/2017 03:30 AM, Sergey Basov wrote: Hi, All. May be it helps. After patch https://github.com/kamailio/kamailio/commit/8ca410cba540e8c8b0f711fb26c85823375480a9 when running kamailio with debug level=3 when I do disable rtpengine I got in log: Feb 6 10:24:27 csbc-uat /usr/sbin/kamailio

Re: [SR-Users] RTPEngine disable/enable crash kamailio

On 02/02/2017 06:54 AM, Sergey Basov wrote: Hello Daniel, You can find backtrace in attaced file Hi, Can you please with the following patch https://github.com/kamailio/kamailio/commit/8ca410cba540e8c8b0f711fb26c85823375480a9 from branch https://github.com/kamailio/kamailio/tree/rfuchs/4.4-rtp

Re: [SR-Users] RTPEngine disable/enable crash kamailio

On 02/02/2017 06:54 AM, Sergey Basov wrote: Hello Daniel, You can find backtrace in attaced file Can you also share the output of "info locals" and "print *node" please Thanks ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing li

Re: [SR-Users] rtpengine - Key file error

On 25/01/17 06:44 PM, Serhat Guler wrote: Hello, I am trying to setup my Kamailio environment on a new Debian system. Everything went well, except I started facing a problem with the rtpengine. It did install it fine and I can view the program parameters via the help menu; however, when I run

Re: [SR-Users] Why SDPOPS does not remove attributes in SDP

On 17/06/16 03:46 AM, Dmitry wrote: Hi all I have the following code: if($T_reply_code=="200") { if(has_body("application/sdp")) { xlog("L_INFO", "RTPENGINE received internal reply $T_reply_code $rr SDP extra lines will be removed");

Re: [SR-Users] Dynamic advertised ip for rtpengine/rtpproxy

On 06/01/2016 04:26 AM, Serge S. Yuriev wrote: Hello, Thanks a lot. If I understand docs correctly there is no such thing for RTPEngine and we should use app params? Yes there is, you can give it as "media-address=..." inside the options string to the respective function calls. Cheers

Re: [SR-Users] Browser WebRTC transcoder

On 05/19/2016 04:52 AM, Moacir Ferreira wrote: ... > So the Grandstream offers a lot of codecs but will get a "Not Found" > from Kamailio. Look in the other way: That would be a SIP signalling (e.g. Kamailio config) problem. Perhaps a missing registration. > Here the Grandstream says "Media type

Re: [SR-Users] Browser WebRTC transcoder

On 18/05/16 04:57 PM, Moacir Ferreira wrote: Hey Daniel, If you say so, you probably right... I did not try it because on the sipwise GitHub (https://github.com/sipwise/rtpengine) they mention: /"Rtpengine does not (yet) support:/ // * /Repacketization or transcoding/ This refers to trans

Re: [SR-Users] question about RTPengine

On 05/06/2016 07:36 AM, Dmitry wrote: > hello > > i read the documentation about RTPengine. > and the documentation says that: > > If INVITE with SDP, when the tm module is loaded, mark transaction with > internal flag FL_SDP_BODY to know that the 1xx and 2xx are for > rtpengine_answer() > > wh

Re: [SR-Users] making rtpengine gives error on centos

On 04/05/16 11:16 AM, Yasin CANER wrote: Hello; i tried to make rtpengine.so deamon that gives error for evthread_use_pthreads is implicit. You need to have a recent libevent installed - libevent-devel or libevent2-devel. http://rpm.pbone.net/index.php3/stat/3/srodzaj/1/search/libevent2-

Re: [SR-Users] about rtpengine versions

On 05/04/16 08:03 AM, Juha Heinanen wrote: I noticed that 15 days ago there was rtpengine commit https://github.com/sipwise/rtpengine/commit/aac8899b612dc8103b89f3f9c921f88af3501303 titled "Release new version 4.4.0.0+0~mr4.4.0.0", but I could not find the corresponding branch or tag. If someo

Re: [SR-Users] RTP Engine multiplexing and De-multiplexing

On 01/13/2016 04:26 AM, riko nir wrote: Hello, A call from a remote webrtc client is coming to (opensips+rtpengine). The media streams from the webrtc client is multiplexed. Can I use rtpengine to demultiplex the multiplexed streams and send it to other end as de-multiplexed SRTP traffic . This

Re: [SR-Users] Regarding rtpengine dtls handling and fetching crypto keys

On 01/13/2016 02:37 AM, riko nir wrote: Hi, Thanks for the answer. Do you have any options for sending this keys to opensips somehow, by modifying the code in rtpengine and in opesips script file? I don't know much about Opensips and so can't provide guidance about how to pass these values ba

Re: [SR-Users] Regarding rtpengine dtls handling and fetching crypto keys

On 01/12/2016 04:09 AM, riko nir wrote: Hi all, I have a media server and it is able to handle SRTP, provided the crypto key. We are planning to give webrtc support to the media server. We are using opensips+rtpengine for that. For dtls, we are using rtpengine. The rtpengine just needs to do t

Re: [SR-Users] RTPPorxy -> RTPEngine migration issue

On 10/01/16 12:48 AM, Arsen Hovhanissian wrote: Hi Richard! Here’s the output at log level 7 [1452404737.610730] WARNING: Failed to properly parse UDP command line '11683_0 d7:command4:pinge' from 10.0.0.10:54602, using fallback RE [1452404737.619006] WARNING: Failed to properly parse UDP comma

Re: [SR-Users] RTPPorxy -> RTPEngine migration issue

On 09/01/16 07:28 PM, Arsen Hovhanissian wrote: Hi everyone, i’ve been having this “not so problem” going on. So I have rtpengine installed on a server and use the default rtpproxy module on kamailio and it works beautifully. Having read the rtpengine modules description, I see that it is a d

Re: [SR-Users] Too many packets in rtpengine UDP receive queue

On 12/23/2015 09:08 AM, Zodiac wrote: Thanks for your analysis. Actually my rtpengine daemon is running on a Virtual Machine. So can there be solution to this problem? Yes, run it on real hardware :) Cheers ___ SIP Express Router (SER) and Kamailio

Re: [SR-Users] Too many packets in rtpengine UDP receive queue

On 12/22/2015 10:21 AM, Zodiac wrote: Hi friends: I am running rtpengine daemon on a CentOS machine functionally. SIP server is Kamailio. Vedio and audio calls are both OK with rtpengine. But there is always prompts like following in rtpengine’s log: Dec 22 19:57:52 localhost rtpengine[12679]:

Re: [SR-Users] why is rtpenengine-delete deleting the whole call?

On 12/04/2015 07:19 PM, Juha Heinanen wrote: Richard Fuchs writes: question: how it is possible the call works, i.e., rtpengine still has the call even when it was already deleted? does it remember that two offers were made using same params and the call does not really get deleted before it

Re: [SR-Users] why is rtpenengine-delete deleting the whole call?

On 12/03/2015 06:38 PM, Juha Heinanen wrote: i noticed one more things during testing of rfuchs/delete-branch, which i don't quite understand. the test call is parallel forked to two destinations and offer (using via-branch=1) is issued for both. thus the offers have the same params: ... "call

Re: [SR-Users] why is rtpenengine-delete deleting the whole call?

On 11/27/2015 07:41 PM, Juha Heinanen wrote: Richard Fuchs writes: To clarify: the extra-pv parameter simply replaces the via-branch value with a custom string. Got it and when standard via-branch value is taken into account that would be enough for me at least for now. I have a

Re: [SR-Users] why is rtpenengine-delete deleting the whole call?

On 11/26/2015 09:21 PM, Juha Heinanen wrote: Richard Fuchs writes: question: why does rtpengine not do what it is asked to do in rtpengine-delete, i,e. delete the offer corresponding to "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd

Re: [SR-Users] why is rtpenengine-delete deleting the whole call?

On 11/26/2015 07:39 PM, Juha Heinanen wrote: question: why does rtpengine not do what it is asked to do in rtpengine-delete, i,e. delete the offer corresponding to "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1"? Because the via-branch isn't take

Re: [SR-Users] rtpengine via-branch=extra question

On 11/22/2015 05:50 AM, Juha Heinanen wrote: Richard Fuchs writes: Yes, one of the two usual call timers. Off the top of my head I believe the longer ("silenced") timer is the relevant one here. Just for curiosity, I tried to find out from the sources, how long that SILENT tim

Re: [SR-Users] rtpengine via-branch=extra question

On 11/19/2015 05:58 PM, Juha Heinanen wrote: Richard Fuchs writes: if via-branch=extra is used in offers of branched, should it also be used in answers and deletes? Yes it should, as rtpengine uses the via-branch to match the answers to the offers which didn't have a to-tag. Looking a

Re: [SR-Users] rtpengine via-branch=extra question

On 11/18/2015 08:47 AM, Juha Heinanen wrote: rtpengine readme tells: via-branch=... - Include the “branch” value of one of the “Via” headers in the request to the RTP proxy. Possible values are: “1” - use the first “Via” header; “2” - use the second “Via” header; “auto” - use the fir

Re: [SR-Users] rtpengine disabling offers of rtcp

On 08/10/2015 05:45 AM, Daniel Tryba wrote: > On Wednesday 05 August 2015 16:05:52 Daniel-Constantin Mierla wrote: >> maybe you can just removed with textops or sdpops functions plus >> msg_apply_changes(). > > Somehow that doesn't work for me... The a=rtcp line remains > whatever/whereever > I

Re: [SR-Users] RTPengine+Kamailio, 200 OK without DTLS fingerprint

On 02/06/15 06:35 AM, Vasiliy Ganchev wrote: > Richard Fuchs wrote >> On 29/05/15 11:16 AM, Vasiliy Ganchev wrote: >>> On May 29, 2015; 3:19pm, Richard Fuchs wrote: >>>> A good way to start debugging this is to run rtpengine with log-level 7 >>>> and

Re: [SR-Users] RTPengine+Kamailio, 200 OK without DTLS fingerprint

On 29/05/15 11:16 AM, Vasiliy Ganchev wrote: > On May 29, 2015; 3:19pm, Richard Fuchs wrote: >> A good way to start debugging this is to run rtpengine with log-level 7 >> and post the full log for such a call. > Hi Richard! Thanks for answer! > Call log written on WS_Kam

Re: [SR-Users] RTPengine+Kamailio, 200 OK without DTLS fingerprint

On 29/05/15 05:48 AM, Vasiliy Ganchev wrote: Юрий, Thanks for your answer! Sorry but I couldn't understand how to do "Reverse rtp_manage functions". If it is possible, extend your idea with some more example/explanation? Maybe someone else can help suggest something, or do I need to give more in

Re: [SR-Users] rtpproxy-ng and late SDP

On 18/05/15 03:53 AM, Sebastian Damm wrote: Hi Alex, On Thu, May 14, 2015 at 5:47 AM, Alex Balashov mailto:abalas...@evaristesys.com>> wrote: According to the rtpengine module documentation for rtpproxy_manage(), that's exactly what rtpproxy_manage() does: http://kamailio.org/doc

Re: [SR-Users] rtpengine, turn fallback

On 06/05/15 01:30 AM, Isuru Wijesinghe wrote: Hi All I'm using rtpengine to in my implementation of webrtc. In my client side browser I have used following code snippet to configure stun/turn servers to identify ice candidates and turn fall back. >>

Re: [SR-Users] rtpengine and security

On 21/04/15 10:40 PM, GG GG wrote: > By port closed, I mean that ports are normally closed, but when > rtpengine send the first rtp packets to the client, it opens a pinhole > in the firewall, and the matching incoming packets from the client will > make the connection established,related in iptabl

Re: [SR-Users] rtpengine and security

On 21/04/15 11:04 AM, GG GG wrote: > Hello, > > what do you think about opening all RTP ports for rtpengine on Internet, > is it a bad practice ? > > I wonder if it's possible to use rtpengine with all ports closed. Not sure what you mean with "ports closed." How would rtpengine, or any other RT

Re: [SR-Users] weight of rtpengine rtp proxy?

On 08/04/15 04:19 AM, Juha Heinanen wrote: > rtpengine readme tells in section 2: > > The balancing inside a set is done automatically by the module based on > the weight of each RTP proxy from the set. > > i have not found in rtpengine_sock module parameter description, how is > the weight o

Re: [SR-Users] rtpengine documentation error for forced rtpproxying?

On 01/04/15 12:34 PM, Daniel Tryba wrote: > On Wednesday 01 April 2015 18:08:28 Daniel Tryba wrote: >> rtpengine_manage("force"); > > Whoops, I should read the whole thing. > > http://kamailio.org/docs/modules/4.2.x/modules/rtpengine.html#rtpengine.f.rtpengine_offer > > force - instructs the RTP

Re: [SR-Users] RTPEngine IPv4 to IPv6 bridging returning c=IN IP4 0.0.0.0 on answer?

On 24/02/15 08:20 PM, Anthony Messina wrote: > This is probably very likely a configuration issue on my part, but I wanted > to > check before reporting an RTPEngine bug... > > Thank you for any pointers or suggestions. > > This is a multi-homed server where > > em1: INTERNAL_IPv4 & GLOBAL_IPv

Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

On 16/02/15 01:12 PM, Virmantas Variakojis wrote: > Could you provide us a little example? For examlple i have kamailio with > three interfaces: two interfaces (vlan's look at two different > providers) and third interface looks at sip clients. You would define two interfaces with different names,

Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

On 16/02/15 01:00 PM, Virmantas Variakojis wrote: > Hi, > > There pathch with -A can be found or it is allready implemented into > specific rtpengine version? Latest master from git. The command line syntax is a bit different from rtpproxy, but the basic idea is the same. Cheers ___

Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

On 16/02/15 12:39 PM, Muhammad Shahzad wrote: > I haven't done something like that myself but i think if you use > RTPEngine with "media-address" set correctly in offer and answer > functions, you can easily achieve this. Simply check if request/reply is > coming from FS or the end-user and adjust

Re: [SR-Users] rtpengine - Error when sending message. Error: Invalid argument

On 14/02/15 08:13 AM, Amit Patkar wrote: > Hi > > I am getting error with rtpengine. > Running Kamailio 4.2.3 > > I am trying to call from conventional SIP client to WebRTC client > > Google Chrome v38.0.2125.104 > Firefox v33.0 > > Using sipML5 as WebRTC client > > root@rtcpbx:/home/avhan# /u

Re: [SR-Users] Kamailio and rtpengine

On 13/02/15 01:32 PM, Marc Soda wrote: > How does Kamailio load balance traffic to rtpengine? Is it load based, > round robin, etc? The module makes mention of this but I don't see how > it works. Also, it talks about weighting the proxies. How is that > accomplished? This part of the module i

Re: [SR-Users] One Way RTP from destination SIP server using rtpengine

On 07/02/15 07:24 AM, Muhammad Shahzad wrote: > I think i have similar problem last week with rtpengine deployment which > was about 1-2 weeks old. There was no audio although the logs say that > STUN bindings are successful from both side (SAVPF <-> AVP). One symptom > of the problem is this log m

Re: [SR-Users] RTPEngine Intermittent One Way Audio Issue SRTP => RTP

On 07/02/15 03:09 AM, Tim Chubb wrote: > Hi, > > > > I am currently experiencing an intermittent one way audio issue, when > using RTPEngine and proxying between srtp and rtp. There is no > apparent pattern, I have managed to repeat the on the 7^th , 23^rd , > and 63^rd calls to a test numbe

Re: [SR-Users] Video Key-Frame Request using RTCP FIR or SIP INFO message

On 02/01/15 09:17, Muhammad Shahzad wrote: > Thanks for detailed reply and sharing the valuable information. You are > right i should also post this on discuss-webrtc forum, will do after i > get a fresh call trace, possibly tomorrow. > > Regarding your questions, yes the call establishes successf

Re: [SR-Users] Need help on WebRTC with Kamailio as proxy

On 26/01/15 02:21 PM, Rahul MathuR wrote: Hello, I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progres

Re: [SR-Users] RTPengine

On 01/19/15 08:11, Kalala Alexander wrote: > Is there a replacement of this flag "/force/"? Rtpengine is always in "force" mode. It always tries to proxy the media through itself when instructed to. If you need to prevent the media from being proxies twice, you need to find some other way to detec

Re: [SR-Users] rtpengine stats

On 12/22/14 10:33, Marc Soda wrote: > Does anyone know how a can get stats from rtpengine? I see the $rtpstat > pseudo variable in Kamailio, but from the documentation it looks like > that will only give me stats on a particular call. I'm looking for > overall stats like concurrent calls, bandwid

Re: [SR-Users] Media trouble with kamailio/rtpengine

On 12/19/14 10:47, Marc Soda wrote: > I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not > getting audio back to my browser. From a packet capture I can see media > from the browser to rtpengine, and then bi-directional RTP back and > forth from my asterisk server, but rtpengin

Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer

On 12/19/14 11:39, Juha Heinanen wrote: > Richard Fuchs writes: > >> I don't see how it would make a difference. If Firefox sends 0.0.0.0 and >> rtpengine replaces it with its own address, then the receiving client >> can send media to rtpengine, but rtpengine would

Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer

On 12/19/14 10:02, Juha Heinanen wrote: > Richard Fuchs writes: > >> Yes I understand, but 1) the mechanism of using 0.0.0.0 to put a call on >> hold must remain operational and intact for those clients which use it, >> and 2) if the offering client sends 0.0.0.0 in the SD

Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer

On 12/19/14 09:33, Juha Heinanen wrote: > Richard Fuchs writes: > >> On 12/19/14 03:32, Juha Heinanen wrote: >>> i got mozilla to generate sdp with sendrecv, but still rtpengine does >>> not replace 0.0.0.0 address on o and c lines. why? >> >> Because 0.

Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer

On 12/19/14 03:32, Juha Heinanen wrote: > i got mozilla to generate sdp with sendrecv, but still rtpengine does > not replace 0.0.0.0 address on o and c lines. why? Because 0.0.0.0 means steam is on hold and so should be left in place. cheers ___ SIP

Re: [SR-Users] webrtc clients support using rtpengine

On 12/18/14 13:38, Andrey Utkin wrote: > This works: call from sipml to linphone android: > rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58 > kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46 > ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3 > > >

Re: [SR-Users] webrtc clients support using rtpengine

On 12/18/14 12:55, Andrey Utkin wrote: > 2014-12-18 19:30 GMT+02:00 Richard Fuchs : >> Write error on RTP socket usually indicates an incorrect network setup, >> for example trying to use a source address for IP packets which isn't >> bound to any local network inter

Re: [SR-Users] webrtc clients support using rtpengine

On 12/18/14 12:11, Andrey Utkin wrote: > Hi! > I need to establish calls between WebRTC and usual SIP clients > (exactly, sipml/jssip and linphone-android). > I used configs from https://github.com/caruizdiaz/kamailio-ws and > latest git master HEAD of both kamailio and > rtpengine. I got calls fro

Re: [SR-Users] SIP Fragments

On 12/17/14 21:14, Alex Balashov wrote: > And yeah, WebRTC is morbidly obese. I think your best bet is to use TCP. ... or try to fix your networking. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip

Re: [SR-Users] How to Fix the RTP Engine SDP Rewriting direction.

On 12/16/14 10:04, Mahmoud Ramadan Ali wrote: > Hi Dears, > I'm working on integrating the rtpengine to work with Kamailio as RTP > proxy and i have successfully configured the rtpengine in multi home > mode to proxy the media and rewrite the SDP message whenever it passes > trough the Kamailio int

Re: [SR-Users] Error when configuring RTPProxy-NG.

On 12/14/14 18:08, Mahmoud Ramadan Ali wrote: > Hi Dears, > I'm following this link https://github.com/sipwise/rtpengine to > configure rtpproxy-ng with Kamailio BUT i'm reciving the following > errors in the Kamailio log file. > > Dec 14 17:58:34 debian /usr/local/sbin/kamailio[6198]: ERROR: > rt

Re: [SR-Users] rtpengine use of udp

On 05/12/14 04:38 AM, Juha Heinanen wrote: what has been the motivation that rtpengine control protocol is using udp instead of tcp or ssl? we have noticed that reply to delete containing stats is too large for a single udp packet over ethernet and causes udp fragmentation and fragmentation need

Re: [SR-Users] rtpengine UDP/TLS/RTP/SAVP <=> RTP/AVP re-INVITE

On 11/18/2014 08:29 PM, Juha Heinanen wrote: > Richard, > > Enclosed find syslog that includes rtgpengine log at level 7 and also > pcap of the call. I made re-invite from baresip to sems once I saw this > in syslog: > > Nov 19 03:22:14 rautu rtpengine[29718]: [5315d797a9a5e7ce port 50761] > DT

Re: [SR-Users] rtpengine and sdes

On 11/19/2014 01:35 AM, Daniel-Constantin Mierla wrote: > Hello, > > I saw that rtpengine docs still advertise support for SDES SRTP and I > was wondering if anyone was (is still) using it for > decryption/encryption of this type of SRTP, mainly in scenarios like > SRTP from a classic sip phone (e

Re: [SR-Users] rtpengine UDP/TLS/RTP/SAVP <=> RTP/AVP re-INVITE

On 11/17/2014 08:03 PM, Juha Heinanen wrote: > when i make call from UDP/TLS/RTP/SAVP baresip to RTP/AVP sems, > rtpengine gets called on initial invite/200 ok like this and audio works > fine: > > Nov 18 02:46:06 rautu /usr/bin/sip-proxy[926]: INFO: = > rtpengine_offer(ICE=force replace-sess

Re: [SR-Users] rtpengine flags UDP/TLS/RTP/SAVP and UDP/TLS/RTP/SAVPF

On 11/17/2014 08:14 AM, Juha Heinanen wrote: > Juha Heinanen writes: > >> rtpengine source code, however, seems to know also these DTLS ones: >> >> daemon/call.c: .name = "UDP/TLS/RTP/SAVP", >> daemon/call.c: .name = "UDP/TLS/RTP/SAVPF", >> >> can th

Re: [SR-Users] rtpengine does not send RTP packets out

On 11/03/14 09:17, Juan Perez wrote: > thank you Richard, yes the IP is local to the machine: > > ./rtpengine --interface=pub/ --interface=priv/10.0.2.68 > --listen-ng=127.0.0.1:7722--timeout=30 --port-min=35000 --port-max=65000 > --log-level=7 --log-facility=daemon > > The PUBLIC_IP is a NAT tha

Re: [SR-Users] rtpengine does not send RTP packets out

On 11/01/14 15:39, Juan Perez wrote: > Hi, I have kamilio-4.2 and rtpengine running on the same machine. > I have this scenario: > > softphone --> Kamailio with Rtpengine --> Asterisk > The softphone initiates the call, it is sent to the Asterisk. I can see > the SDPs being re-written with the new

Re: [SR-Users] Installing rtpengine on CentOS 6.5

On 10/31/14 17:31, Juan Perez wrote: > Thank you, but after removing the "-e" now I get this error: > > [root@ip-10-0-2-68 daemon]# make > Makefile:88: .depend: No such file or directory > make: *** No rule to make target `aux.c', needed by `.depend'. Stop. > [root@ip-10-0-2-68 daemon]# Are you

Re: [SR-Users] Installing rtpengine on CentOS 6.5

On 10/31/14 15:27, Juan Perez wrote: > > Using Kamailio 4.2, CentOS 6.5, as per the documentation here: > https://github.com/sipwise/rtpengine > > "Running make will compile the binary, which will be called rtpengine." > > When trying to install rtpengine I got the following error when building

Re: [SR-Users] (no subject)

On 10/29/14 10:08, Yuriy Gorlichenko wrote: > Hello. I use kamailio with last rtpengine and > I have 5-7 Seconds voice delay. This happened only for from webphone. > But it is not client issue as i see. Wireshark at client side shows that > RTP starts as soon I pick up call. So rtp leaves rtpengi

Re: [SR-Users] rtpproxy_manage fails to rewrite SDP

On 10/28/14 09:18, Igor Potjevlesch wrote: > Hello, > > > > I have seen that the problem can occur due to the kernel limitation. > > So, I have changed the local_port range to be sure that they include the > port range of RTPProxy. > > > > But, I still not explain this limitation. Because,

Re: [SR-Users] Kamailio for ARM with rtpproxy-ng

On 10/27/14 06:57, Marino Mileti wrote: > Richard Fuchs wrote >> Interesting. I'm not sure why this is required for this value, but hey, >> if it works... Let me know if you run into any more trouble. > > Bad news..the __attribute__((packed)) modification allows only

Re: [SR-Users] Kamailio + RTPEngine behind NAT (STUN Handling)

On 10/24/14 12:34, Marko Seidenglanz wrote: > Hello, > > We want to use Kamailio (4.2) + RTPEngine (3.3) behind NAT. > Unfortunately the STUN Messages (Binding Requests) sent from remote peer > (Google Chrome ) do not arrive at RTPEngine host. > > I think they get blocked by the firewall. RTPEngi

Re: [SR-Users] rtpengine not working with append_branch

On 10/23/14 15:06, Yuriy Gorlichenko wrote: > Still have same error... > Now rtpproxy_manage("co-sp") for classic call. At log I see that > rtpproxy wirked gine. For each step it generate write body, but t_Relay > still send strange "compinated" packet to UDP with SDP for WS... Do you mean that t

Re: [SR-Users] rtpengine not working with append_branch

On 10/23/14 12:17, Yuriy Gorlichenko wrote: > What you mean under "full set of flags"? At reply I use mirror (+/-) > flags off course. More, it work without branches fine ( i select only > one endpoint). I have issue only with branches. I mean that instead of using rtpproxy_manage("co") you should

Re: [SR-Users] Kamailio for ARM with rtpproxy-ng

On 10/23/14 11:32, Marino Mileti wrote: > Version is 3.3.0.0 mr3.5.0.0 > > I've seen that the problem is during the "ping-pong test" when the value of > bencode_item is assigned > > The error is on __bencode_dictionary_init, and the istruction is : > dict->value=0 > > I've read ARM documentation

Re: [SR-Users] rtpengine not working with append_branch

On 10/23/14 06:03, Yuriy Gorlichenko wrote: > Hello all. I use rtpengine and rtpproxy-ng module at kamailio for > proxying RTP and modifying SDP between endpoints. I use two types of > clients - such as WSS based and UDP based clients. > > I have a trouble with append_branch and rtpengine handling

Re: [SR-Users] Kamailio for ARM with rtpproxy-ng

On 10/23/14 05:28, Marino Mileti wrote: > Hi guys, > > I've cross compiled Kamailio (4.1.6) for ARM (Wandboard-IMX6). Everything > seems works fine, but when I try to enable rtpproxy-ng module and Alignment > trap error occurred just during rtpp_test (exchanging ping-ponge message > betweek rtppro

Re: [SR-Users] rtpengine upgrade fails

On 10/14/14 04:08, Daniel-Constantin Mierla wrote: > > On 13/10/14 18:07, Juha Heinanen wrote: >> just to complete the thread, the dependency problem has been fixed and >> upgrade from 3.3.0.0+0~mr3.5.0.0 to 3.3.0.0+0~mr3.6.0.0 succeeded >> without issues. >> > Is it still possible to install the

Re: [SR-Users] how to always use rtpproxy-ng in kamailio

On 10/05/14 07:15, andrew wrote: > Hi, > > I have one question about kamailio. > > rtpproxy-ng is installed with kamailio. When client A initiates one call > to client B, SDP in invite/200ok are rewritten, where ice candidates for > rtpproxy-ng are added. Sometimes rtp packets are relayed throug

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On 09/29/14 14:51, Marino Mileti wrote: > 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 s

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

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 t

Re: [SR-Users] rtpproxy & extra_id_pv

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 __

Re: [SR-Users] rtpproxy & extra_id_pv

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 umbrel

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On 09/29/14 13:29, Frank Carmickle wrote: > > On Sep 29, 2014, at 1:24 PM, Richard Fuchs wrote: > >> On 09/29/14 13:19, Frank Carmickle wrote: >>> >>> On Sep 29, 2014, at 1:14 PM, Richard Fuchs wrote: >>>> >>>> This may work with rtpeng

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On 09/29/14 13:19, Frank Carmickle wrote: > > On Sep 29, 2014, at 1:14 PM, Richard Fuchs wrote: >> >> This may work with rtpengine, as it will open new ports for answers come >> from different endpoints. But the final two-way association for the >> actual call may

Re: [SR-Users] rtpproxy & extra_id_pv

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("rtpp

Re: [SR-Users] rtpengine with rejected re-invites to new RTP ports

On 09/25/14 12:05, Jeff Pyle wrote: > Hello, > > Given the following scenario with Kamailio and rtpengine in the middle: > > - call establishes with G.711 RTP > - b-leg re-invites to T.38, indicating a different port number then he > is using for G.711 > - a-leg refuses the re-invite with a 48

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On 09/25/14 10:41, Marino Mileti wrote: > > 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 r

Re: [SR-Users] R: Re: RTPPROXY & BRANCH

On 09/29/14 13:03, Richard Fuchs wrote: > On 09/25/14 10:22, Frank Carmickle wrote: >> >> On Sep 25, 2014, at 10:09 AM, Marino Mileti > <mailto:marino.mil...@alice.it>> wrote: >> >>> Because I've more than 1 client behind NAT (1,2,3 mobile phones

Re: [SR-Users] R: Re: RTPPROXY & BRANCH

On 09/25/14 10:22, Frank Carmickle wrote: > > On Sep 25, 2014, at 10:09 AM, Marino Mileti > 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 >>

Re: [SR-Users] rtpproxy_offer and fix_nated_sdp in one route

On 09/24/14 09:16, Sebastian Damm wrote: > 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

Re: [SR-Users] RTPPROXY & BRANCH

On 23/09/14 04:35 AM, marino.mil...@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 po

Re: [SR-Users] rtpengine upgrade fails

On 08/21/14 03:09, Juha Heinanen wrote: > Now I made another dist-upgrade upgrading rtpengine from > 3.3.0.0+0~mr3.4.1.0 to 3.3.0.0+0~mr3.5.0. Indeed dist-upgrade tries to > setup daemon after the kernel module was removed but before new kernel > module was installed. ... > Setting up ngcp-rtpengi

Re: [SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

On 08/25/14 19:25, Alex Villací­s Lasso wrote: > I have a rtpproxy configuration that spawns several rtpproxy instances, > using bridge mode. An example is shown below: > > /usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s > udp:127.0.0.1 7723 192.168.2.18/127.0.0.1 -m 1 -M 2

Re: [SR-Users] OT: RTCP logging

On 09/04/14 12:01, Daniel Tryba wrote: > On Thursday 04 September 2014 15:22:52 Daniel-Constantin Mierla wrote: >> - http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp1673992 >> >> rtpengine (former rtpproxy-ng) should have it as well, I guess. > > Found this before posting, but I c

Re: [SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

On 08/25/14 19:25, Alex Villací­s Lasso wrote: > I have a rtpproxy configuration that spawns several rtpproxy instances, > using bridge mode. An example is shown below: > > /usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s > udp:127.0.0.1 7723 192.168.2.18/127.0.0.1 -m 1 -M 2

Re: [SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

On 08/26/14 20:58, Alex Balashov wrote: > On 08/26/2014 08:56 PM, Paul Belanger wrote: > >> I'd agree 'drop-in' replacement is not correct. I ran into the same >> issues as you. Current there is no bridge-mode in rtpengine, I point >> you to an open issue about it [1]. > > I think the idea behind

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