search for: rtpproxi

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

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 ?

hi, hs 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

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

You will also need your SIP clients that are behind the same NAT to support ICE (Interactive Connectivty Establishment) if you want calls between them. Xten Eyebeam and Snom phones are the only ones I'm aware of that support it. On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check

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

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.xxx disallow=all allow=ulaw

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

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:

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

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.

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

Hi, I would need to bridge a SIP call with a multicast RTP channel. Both sides are receiving and transmitting RTP. 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

Hello, http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly The link above indicates 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

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

Hello, ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as 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

2017-02-16 14:27 GMT+01:00 Joshua Colp <jcolp at digium.com>: > On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote: > > Hello, > > > > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP > > hardphone. > > > > When a phone has enabled this feature, it would send a SIP PUBLISH to its > > SIP Server letting this server dispatch to

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....