similar to: OT: RTP - NAT - SBC

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

Dear list: I have the typical one way audio problem, as far as i know it's a nating problem, my hosts inside my lan can call to outside internet hosts, but can't listen a thing, i read a lot about sip and rtp and protocols and the problem it seems to be with NAT, this is the config i put on my sip.conf file about nat: externhost=sip.server.com.ar > my server name on the

I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA -----> Asterisk ----> SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP

Hello List Asterisk 13.14.1 in use with pjsip stack. On the remote side is a SBC which performs some 'nat' detection. I suppose this means the SBC listens from where it is getting RTP data and then replies to that ip. As long as the asterisk is initiating the call this is fine, the asterisk start sending RTP to the media IP of the SBC and the SBC is sending media back. Now I want to do

Hello I think I finally understood the issue/solution, but I'd like to make sure I'm correct: - In Diana Cionoiu's famous article on Freshmeat (http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol), regardless of whether SIP end-points use a public IP or are behind a NAT, RTP packets flow directly between the two SIP end-points because the SIP server only acts... as an

http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is working fine in my Setup, but I want Extern1 to talk to Extern2 directly whitout going over Asterisk as the uplink is slow. When I set for Extern1/2 canreinvite=yes it works, but "Intern-2-Extern" doesn't work because Asteisk gives out the private IP-Adresses of Int1/2 I defined localnet=10.0.0.0/255.0.0.0 (Private

Hi all, I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the phone is ringing, but when I pickup the call, there's no audio on both sides. I debugged the rtp-traffic at home. As long as the phone is ringing, everything is fine. But after the pickup, asterisk sends a SIP/SDP package with its

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Because it seems my mail from 30th august didn't make it to the list i send it again. If the mail _did_ get to the list and i didn't see it please excuse the duplicate post Below is the mail from the 30th: I have a setup like this: An asterisk-server with SIP-phones on the outside of a NAT. For example: asterisk with local IP-address

*Hi All, I have starting to reading About SBC and found one artical reagding SBC and they gives a solutions like this. i want to know is this true in realtime sceanario while we think of an big implementation and is it possible with cloud computing. i have found from http://www.smartvox.co.uk/products_gateways_explained.htm Asterisk as a Session Border Controller* Equip the Asterisk server

I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress on we are having a few disconnects while calls are in session. I have talked both to some local phone contractors and SBC directly and

I have been studying Tom''s configuration at: http://www.shorewall.net/myfiles.htm -and- http://www.shorewall.net/NAT.htm I am using SBC as an ISP and also have 5 "real" IP addresses and because of other issues, have to re-do my set-up. If I have a block at .120/29 assigned to me, what SBC does is give you 5 usable addresses, in my case .121 is the SBC modem/router and

Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM]

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

Hi, I'm connecting Asterisk v. 1.4.10 to Zanzibar Open IVR that acts as a SIP trunk. Since recognition didn't work correctly, I've troubleshot with Wireshark and saw that RTP stream is first send to one port on SIP trunk and then when first RTP packet arrives in opposite direction (from TTS part of Zanzibar - it's a prompt) Asterisk starts sending to the same RTP port -

Hi, I am new to the world of Asterisk PBX. I have been given the task of coming up with a solution to our office phone situation. From what I have been reading, Asterisk sounds like it could be ideal. However, most of the information I am finding is focused on the VoIP aspects, which is something we want to use for our remote employees, but the first hurdle is integrating Asterisk with the

Good day. I'm trying to configure termination with The Asterisk thru Cisco AS5300 Gateway from the SIP softphone (X-Ten X-Lite) to POTS network. I think, I had recognise kind of problem: call is ringing in the POTS phone (so I guess SIP signalling is working ok?), but there is no voice in either sides. On the Asterisk PC I can see incoming RTP streams with tcpdump and tethereal, but I

Hi! I have this configuration: SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real IP) <-> (real external IP) NAT box B <-> SIP client B The echo test form any of the clients to the asterisk server is working just fine, even without canreinvite=no. When I try to call from SIP client A to B, wihtout the canreinvite=no in the sip.conf, the call

Hi, I am trying to run an Asterisk (1.4.10) server on Ubuntu Linux Fiesty Fawn. The server is behind NAT. I am testing SIP with the X-Lite client from xten. The client is also behind NAT. I realize that this is amongst the worst configurations, but I have been made to believe that it can work... eventually. However, currently SIP call set up seems to go fine, but no media is transferred in