similar to: Asterisk behind NAT Early Media Video

Displaying 20 results from an estimated 8000 matches similar to: "Asterisk behind NAT Early Media Video"

2018 Apr 09
2
Asterisk behind NAT Early Media Video
Yes, media is flowing through Asterisk because both client's are behind different NAT's. Do I need to do something special in the Call Flow? Or anything additional to the pjsip.conf? 2018-04-09 16:50 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote: > > Hello, > > > > I have an Asterisk 15 with PJSIP behind
2018 Apr 09
2
Asterisk behind NAT Early Media Video
My understanding based on Wireshark analysis is that the signaling works (also the recipent phone is displaying the video frame before accepting the call), also the calling phone send video (i see that also via Wireshark) but the recipent phone doesn't get any video from the Asterisk before the call. 2018-04-09 17:02 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9,
2018 Apr 11
2
Asterisk behind NAT Early Media Video
On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote: > I added the bind_rtp_to_media_address=yes on all endpoints but still the > same behaviour. The funny thing is that the G711 audio early media works > and doesn't have that Private IP issue. I was also able to cross check with > chan_sip on Asterisk 15, exactly the same wrong behaviour. See following > capture (PJSIP): As
2018 Apr 09
3
Asterisk behind NAT Early Media Video
wohoo, so if I unterstand it correctly with that patch early media video works over the Asterisk server? In other words the Asterisk server get's able to (process/)forward the early media video stream with that patch? 2018-04-09 17:57 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote: > > My understanding based on Wireshark
2018 Apr 10
2
Asterisk behind NAT Early Media Video
I just noticed, the calling device isn't even sending the early media video stream. It just sends an early media audio stream. Is there propably a change in the signaling needed? (On another P2P SIP Server the early media video works.) 2018-04-10 12:29 GMT+02:00 Benjamin Marty <benjamin.marty at gmail.com>: > Hi Florian > > I already have the external_media_address set in the
2018 Apr 11
2
Asterisk behind NAT Early Media Video
I did a quick check between what I have set and your settings below. You can try the following and see if it helps In your endpoint: bind_rtp_to_media_address=yes With best regards Florian Floimair Innovation - Software-Development - VoIP & DevOps COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 Tel: +43-662-85 62 25 Fax: +43-662-85 62 26 http://www.commend.com Security
2018 Apr 10
2
Asterisk behind NAT Early Media Video
Hi Benjamin! You're obviously using a similar scenario that I have in place for testing. I initially had issues with early media (not only video also audio) as well in that scenario. What I had to do was to additionally set external_media_address=<your external IP> in pjsip.conf Also, as I wrote the patch for early-media video I'd be interested in any feedback from it. ? ? With
2016 Mar 03
3
RTP / NAT question ( pjsip )
Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump on asterisk server showing UDP packet bound for my remote endpoints internal IP: 17:07:57.130212 IP
2008 Jan 08
1
Early media support for Asterisk behind NAT
Hi, I have asterisk 1.4.16 behind a NAT-FW which is using a hosted SIP trunk for PSTN calling. Asterisk is configured to support nat with nat=yes in sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and early media using 183 Session Progress. So If I call a PSTN number which has IVR message played before the call is connected (via 183), those media RTP packets do not reach the
2009 Feb 20
1
SIP Proxy behind NAT talkinf to ASterisk with public IP
Setup is: Asterisk --->NAT--> SIP Proxy I have following entry for SIP Proxy in sip.conf [Proxy] type=peer host=Static IP (NAT Firewalls public IP) username=xxxx secret=xxxxx nat=yes???????????????? canreinvite=no???????? qualify=yes Proxy sends a call and I get this error Found no matching peer or user for <NAT's Public IP:70001 NAT is using 70001 as the source port in the
2014 Dec 05
3
two tinc hosts behind same NAT
Dear all, I have 3 nodes: A, B and C. C has external IP and A and B are behind NAT. It turns out A and B route their traffic via the C, which they ConnectTo with; this instead of getting connection details from one another and contacting eachother directly (mesh style). The reason is, as I conclude from tincd debug output, is that they see the peer as having a minimum MTU of 0. I suspect this is
2015 Sep 25
2
Tinc clients behind a NAT, tunnels get unstable
Hi Guus, Am Freitag, den 25.09.2015, 09:36 +0200 schrieb Guus Sliepen: > On Fri, Sep 25, 2015 at 08:41:06AM +0200, Marcus Schopen wrote: > > > I'm running some tinc clients behind a NAT (masquerading, Cisco Router) > > connecting to a host outside on a public IP in a different network. The > > tunnels get unstable every few minutes and I see packet loss when > >
2003 May 31
0
register with outbound proxy from behind nat for freeworlddialup etc.
Hi, I've posted a simular message little over a week ago so sorry for reposting. I need to register to freeworld dial up from behind a nat. Using the xten software sip client works fine but with asterisk I don't know how to do it. Last time I posted I got different responses. Some saying I can't register with an outbound proxy from asterisk others said they have done it. If it is
2003 Dec 18
8
asterisk behind NAT
I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients
2020 Aug 17
2
Queue don't call Interface PJSIP
Hello. I am having a lot of problems with SIP through NAT. So, I decided to adopt PJSIP. However, I am not able to make the extensions ring when receiving a call from the queue. I'm using telnet to include the extension and on the asterisk console, it even shows Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally. telenet:
2006 Oct 31
2
Asterisk both behind a NAT and outside at the same time
I've read a lot of the descriptions of handling NAT with Asterisk, and the use of both the nat and canreinvite flags. I am very familiar with Sip and NAT but have not seen an answer to the following question. My Asterisk server runs on a machine with two ethernets. One is an external net, with exposed IP addresses. The other is an internal net with natted IP addresses. Thus the server
2014 Dec 05
0
two tinc hosts behind same NAT
Hi Eric, Which version are you using? I have similar issues with the newest 1.1pre10. Did you check out the ?LocalDiscovery? option? For 1.1, you also can get more information about the actual connection mode by using the ?info [node]? command. Cheers, Steffen > Am 05.12.2014 um 12:26 schrieb Eric Feliksik <feliksik at gmail.com>: > > Dear all, > > I have 3 nodes: A, B
2006 Jan 12
0
How to register a SIP phone on Asterisk behind NAT
I currently do this for about 30 different cisco 79xx's connecting to some hosted Asterisk servers. Asterisk listens by default for any SIP connection on UDP port 5060. And will use RTP UDP port 10000 to 20000 The phones use UDP Port 5061 for incoming connections (from Asterisks or other SIP Devices) and use for RTP, UDP port 10000 to 20000. Now, if you are going to have the two remote
2004 Aug 26
0
Asterisk media problem behind NAT
Hello All, I have a media problem while using sip communicator user agent with asterisk behind NAT.I had enabled the debug mode in asterisk and capture the results.I have attached the results with this mail.Can any one help me to fix the problem? Thanks in advance, Partha __________________________________ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out!
2015 Sep 25
2
Tinc clients behind a NAT, tunnels get unstable
Hi Guus, Am Freitag, den 25.09.2015, 17:04 +0200 schrieb Guus Sliepen: > Ok, that means by default the UDP NAT timeout on the Cisco is extremely > short. > > > I check the manual of the the Cisco NAT for any TCP/UDP > > timeout settings, but there is no way to modify anything like "keeps > > TCP/UDP connections alive". > > It wouldn't be called