search for: rtpproxy

Hi asterisk, openser, ser users. I have to check video support between asterisk, open(ser) and rtpproxy . ASTERISK (b2bua+registrar server) | | | | SER + rtpproxy | | NAT | | sip agents (with video support) Both signalling and media channels are kept in the path of SER+rtpproxy and ASTERISK . I can provide a...

Hi All, I've been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not. I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting "directmediadeny|directmediapermit" to offload media from asterisk for peer-to-peer calls BUT what if someone wants to record a call or engage some feature-code ? Will asterisk re-negotiate/re-invite both peers to have Media flowing through asteris...

...s anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start troubleshooting this? TIA Regards nhadie

I have a really simple question about a fairly complex problem: I have a Cisco 7960 behind a NAT. I have an Asterisk server behind a different NAT. I have a SER server (with rtpproxy installed) on a public IP adress. I've opened ports with static NAT to * and the Cisco. Without using SER, I can register the phone to *, I can complete calls, I just can't move audio. Reading the archives and voipinfo seems to indicate that I can use SER and rtpproxy as a "mid...

...ovanni Balasso > > CC: serusers@iptel.org > > Asunto: Re: [Serusers] NAT considerations... > > > > > > Giovanni Balasso wrote: > > > > >Just some thoughts based on my experience... > > >After months trying to make everything work using > > rtpproxy-mediaproxy with > > >almost everything accomplished but video, I tried to switch > > to stun solution. > > >All my problems are gone now, I have audio, video, presence > > and instant > > >messages working like a charm. And most important media > > serve...

Hello, is it possible to set an IP address for RTP different than the one used for SIP? I want to use asterisk behind a sip proxy (opensips), but I was thinking if I could avoid having to run rtpproxy on the sip proxy server and let asterisk itself take care of it. So that: Asterisk SIP address : local ip address Asterisk RTP address : global ip address regards, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/aste...

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx....

Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how they work but am a little stumped by password synchronization using existing LDAP accounts. Maintaining separate accounts with a shared...

Hello I have one phone (phone1) in one network, the other (phone2) in public network. both can call the other side; phone1 can be heard by phone2, phone2 can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER. Is NAT still necessary to be set on both phones? Thank you! steven

Hello, I have asterisk connected to SER/RTPProxy which is again connected to a IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone connected to the IP-PSTN gateway, I get 'empty ACKs': U 192.168.0.173:5060 -> 10.254.254.1:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048. Route: <sip:...

Hello, I wish to set a sip uri sip:info@mydomain. I use ser for authorization and authentication (registrar rtpproxy and outbound proxy) I use asterisk 1.2.5 with realtime . the info is used as a hunt group so i add in extension.conf [info] exten => info,1,Answer() exten => info,n,Dial(Sip/84,10) exten => info,n,Dial(Sip/85,10) exten => info,n,Hangup Ser forward sip:info@mydomain to asterisk but th...

Hi All, We are making a VOIP application for Mobiles (PDA's) and we are using Asterisk for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP router and routes everything to Asterisk. We also have rtpproxy for SER. Our packet delivery from clients (Mobiles, PDA's) is inconsistent and ranges between 10 to 60 ms delay but the average is near to 20 ms. We use SIP. How can we solve this problem, is there any setting at the server end to handle this, as clients have very limited resources we have to m...

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

.... Googling, I saw that an app_rtppage, which was in the SVN for a while and its not there anymore. It did, I think, only partly what I need (it sent from SIP to the mcast ... not the other way around), but it was a start. Any idea how to do this? I also could use ser/opensips/openser/kamailio with rtpproxy (does rtpproxy support this? it would in any case be a complex modification, I think). But my current setup is based on asterisk, so I'd rather not move it from there or install new apps. Thanks a bunch! Cesc ---------- Forwarded message ---------- From: Cesc Santa <cesc.santa at gmail.co...

...cates that it is possible to setup RTP streams to directly flow between endpoints and completely bypass Asterisk. I would like to know if this configuration would work when, a) both endpoints are behind NAT, and/or b) both endpoints don't support same codecs with media flowing through a SIP+rtpproxy server that can do transcoding ? Thanks and Regards, Vikram.

I think I have got * worked out as far as getting users on a small private network talking with each other, but when it comes to the bigger picture about talking between private networks connected by the Internet then I am getting confused about STUN, SER, SIPPROXY, RTPPROXY, etc. Before I start let me make it clear that I am not looking to drop out onto the public telco network anywhere, not at this stage anyway. I see that as a separate issue. I have a number of organisational entities (oe), each of which has their own Internet domain presence (alice.com, bob.com,...

...ured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the outbound proxy for the softphone(Outbound proxy a.b.c.d:9090). Authentication credentials for the softphone match the user registered in asterisk's sip.conf. F.W runs Kamailio and rtpproxy, with Kamailio listening on port 5060. The asterisk server is setup to listen on port 5060. The Firewall(F.W), uses a libnetfilter_queue based program to : (a) Rewrite the destination port 9090 as 5060, and rewrite all other occurrences of 9090 as 5060 in the SIP message, for packets from the...

Hi, I'm trying to set up a "secure" VoIP channel between a Windows softphone client and an Asterisk 1.6... server running with OpenBSD. By "secure" I mean to prevent any man in the middle to reconstitute any vocal exchange nor sender/addressee/any header data/ of the VoIP call (in first step, I would be glad to secure vocal data ans see later for the header...) I had a

...ike box allow me to do this ? > > You could act as you wish on the PUBLISH requests in Kamailio. > This seams easier, for the moment. I think I still need to better understand what are mixed Asterisk-Kamailio architectures main strengths compared to alternatives (Asterisk alone, Kamailio/RTPproxy, ...) but that is another story. Thank you very much for replying. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > __________________...

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....