Re: [SR-Users] RTPPROXY issue and sip to sip calling

2016-01-31 Thread SamyGo
Hi Rehan, No matter which mode you are running rtpproxy in that IP will always be the IP of the machine it is running on. That means that SDP will take that IP once routed to locally subnet A2B servers. As far as the A2B detecting SIP user as online or offline based on DB, I am not too sure

[SR-Users] RTPPROXY issue and sip to sip calling

2016-01-29 Thread Ahmed Rehan
Dear All I m trying to setup kamailio and asterisk in load balancing with a2billing . Currently all of my VMs, one Kamailio and two asterisks are on same subnet . I have started the RTPproxy like below ./rtpproxy -s udp:127.0.0.1:7722 -l X.X.X.153 -m 1 -M 5 -u root root -F -d INFO

Re: [SR-Users] RTPProxy issue?

2015-03-19 Thread Igor Potjevlesch
. I will let you know if I reproduce the issue with the new version. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Maxim Sobolev Envoyé : lundi 9 mars 2015 19:09 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] RTPProxy issue

Re: [SR-Users] RTPProxy issue?

2015-03-09 Thread Maxim Sobolev
) - Users Mailing List *Objet :* Re: [SR-Users] RTPProxy issue? Ah, ok, I see now. I did not realize you guys are using resizer. Which version of the software are you actually using? I.e. is it latest rel_2_0 / master, or some legacy 1.x code? We've done quite some revamping down there, so

Re: [SR-Users] RTPProxy issue?

2015-03-08 Thread Igor Potjevlesch
: samedi 7 mars 2015 09:14 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] RTPProxy issue? Ah, ok, I see now. I did not realize you guys are using resizer. Which version of the software are you actually using? I.e. is it latest rel_2_0 / master, or some legacy 1.x code? We've

Re: [SR-Users] RTPProxy issue?

2015-03-07 Thread Maxim Sobolev
*Envoyé :* vendredi 6 mars 2015 07:44 *À :* Kamailio (SER) - Users Mailing List *Objet :* Re: [SR-Users] RTPProxy issue? Hi Igor, that's bit strange, since the rtpproxy is not checking any of the rtp flags including marker bit. It would help if you can post a tcpdump capture of the streams

Re: [SR-Users] RTPProxy issue?

2015-03-05 Thread Maxim Sobolev
:* RE: [SR-Users] RTPProxy issue? Hello, Thank you. Just to let you know, the RTPProxy is running in bridging mode. Regards, Igor. *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org sr-users-boun...@lists.sip-router.org] *De la part de* Daniel-Constantin Mierla

Re: [SR-Users] RTPProxy issue?

2015-03-05 Thread Daniel-Constantin Mierla
Hello, maybe Maxim (cc-ed) will be able to provide more insights. Cheers, DAniel On 04/03/15 16:59, Igor Potjevlesch wrote: Hello, I discovered an issue related to the handling of timestamp and/or Marker bit with rtpproxy (I use the latest Extension 20081224). The call-flow is

Re: [SR-Users] RTPProxy issue?

2015-03-05 Thread Igor Potjevlesch
: [SR-Users] RTPProxy issue? Hello, maybe Maxim (cc-ed) will be able to provide more insights. Cheers, DAniel On 04/03/15 16:59, Igor Potjevlesch wrote: Hello, I discovered an issue related to the handling of timestamp and/or Marker bit with rtpproxy (I use the latest Extension 20081224

[SR-Users] RTPProxy issue?

2015-03-04 Thread Igor Potjevlesch
Hello, I discovered an issue related to the handling of timestamp and/or Marker bit with rtpproxy (I use the latest Extension 20081224). The call-flow is the following: one UA places a call to A and put this call on hold. Then, the same UA call another number B. Individual streams are ok.

[SR-Users] RTPproxy issue in forwarding scenario

2013-07-08 Thread Sebastian Damm
Hi, we are building a setup where we use an rtpproxy in all cases. This works fine except for one scenario. Caller - SIP(+rtpproxy) - B2BUA - SIP(+rtpproxy) - Called In this case, the B2BUA implements forwarding and sends the call back through our setup. The B2BUA does not send out a 183

Re: [SR-Users] RTPproxy issue in forwarding scenario

2013-07-08 Thread Sebastian Damm
Hi Daniel, sorry, I probably didn't explain the problem correctly. The SIP part is okay, the ports on the rtpproxy are allocated, but bridging the audio doesn't work until both parties actually send at least one RTP packet. Since both streams (the inbound and the outbound call) end up at one of

Re: [SR-Users] RTPproxy issue in forwarding scenario

2013-07-08 Thread Daniel-Constantin Mierla
Hi Sebastian, I understood, the 'r' flag (iirc being the right one) tells to rtpproxy to trust the ip in sdp and send the rtp to it as soon as received from the other side. It is the same case as chaining many rtpproxies. Cheers, Daniel On 7/8/13 2:25 PM, Sebastian Damm wrote: Hi Daniel,

[SR-Users] Rtpproxy issue with connection information

2012-05-09 Thread Openser Kamailio
Hi, i'm currently working with kamailio 3.2 and rtpproxy 1.2.1. Both are set up on the same computer. When rtpproxy adds an SDP to an Invite, it adds two IPv4 addresses in owner/creator session and connection information field with an error, i.e: *Owner/Connection Information (o)*: doubango 1983

Re: [SR-Users] Rtpproxy issue with connection information

2012-05-09 Thread Andreas Granig
Hi, On 05/09/2012 02:40 PM, Openser Kamailio wrote: *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 *172.27.170.984* 172.27.170.98 *Connection Information (c)*: IN IP4 *172.27.170.984* 172.27.170.98 Could it be possible that you're calling rtpproxy_offer() twice? Andreas

Re: [SR-Users] Rtpproxy issue with connection information

2012-05-09 Thread Openser Kamailio
I call rtpproxy_offer() once, but i use also rtpproxy_manage(). When i disable rttproxy_mange(), it works well. Thanks! On Wed, May 9, 2012 at 2:57 PM, Andreas Granig agra...@sipwise.com wrote: Hi, On 05/09/2012 02:40 PM, Openser Kamailio wrote: *Owner/Connection Information (o)*: doubango

Re: [SR-Users] Rtpproxy issue with connection information

2012-05-09 Thread Reda Aouad
I had the same problem when calling mediaproxy twice by mistake. rtpproxy_manage( ) calls implicitely rtpproxy_offer( ). This is the problem. Either you use only rtpproxy_manage once on the INVITE and let it start and terminate the session, or you use rtpproxy_offer, rtpproxy_answer and