search for: rtpengin

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to configure Asterisk to ignore the rtp profile but allow calls to pass with either of those profiles (even though clients might answer w...

...kamailio_v4.2.x-rpms kamailio-xml.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-xmpp.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms Keep in mind that using Kamailio to bridge the signalling is only half of the equation. You must also bridge the media and so the rtpengine module allows Kamailio to interface with the rtpengine ( https://github.com/sipwise/rtpengine) which does that half. In the provided example Kamailio.cfg there isn't any real hardening and it's pretty much purely used as a bridge that would front an Asterisk 1.8 server for webrtc capabili...

...resent all possible RTP transports to peers. 16.28.0~dfsg-0+deb11u2 (I know it's old, but unfortunately Asterisk was removed from debian 'stable' and the version in 'sid' is just broken (opus + voicemail don't work anymore). But I ran into an issue when the peer is running rtpengine: Asterisk offers: a=candidate:H9da13901 1 UDP 2130706431 157.161.57.1 13104 typ host a=candidate:H1054cffa 1 UDP 2130706431 2001:4060:dead:beef::1 13104 typ host a=candidate:He9b56028 1 UDP 2130706431 fe80::5054:ff:fea2:9057 13104 typ host a=candidate:H9da13901 2 UDP 2130706430 157.161.57.1 1310...

Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source. Customers are starting to ask for web solutions and we need to start testing. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

...ou are simply going to use a pre-canned one then sipml5 works pretty well and remembers your settings in localstorage. I haven't used any closed source versions since the above works really well for us. For the server: If you are using Asterisk 1.8 you'll need to front it with Kamailio and rtpengine (or webrtc2sip but I have had stability issues with that). If you are using a more recent asterisk then the webrtc is built in but I haven't used (we use Kamailio and rtpengine to bridge webrtc). If you need the kamailio config I can send it to you (it gets complicated). The rtpengine works v...

...-xml.x86_64 4.2.1-4.1 > @home_kamailio_v4.2.x-rpms > kamailio-xmpp.x86_64 4.2.1-4.1 > @home_kamailio_v4.2.x-rpms > > Keep in mind that using Kamailio to bridge the signalling is only half of > the equation. You must also bridge the media and so the rtpengine module > allows Kamailio to interface with the rtpengine > (https://github.com/sipwise/rtpengine) which does that half. > > In the provided example Kamailio.cfg there isn't any real hardening and it's > pretty much purely used as a bridge that would front an Asterisk 1.8 ser...

...erver is the latest current 11.14.1. When there's an internal call, Asterisk changes the sdp in the INVITE message and handles the rtp nicely, but it does not do so when the call comes from outside. Why not? Instead, it sends back 488 Not acceptable here. If I react to that in Kamailio and use rtpengine to rewrite the sdp, Asterisk accepts the INVITE and sends it to the websocket peer, but the sdp contains a very simple sdp with RTP/AVP profile. This I'd consider invalid behavior, since Asterisk knows the called party is webrtc and the INVITE already contains valid sdp with RTP/SAVPF profile....

...e > kernel, like iptables does, it would take 10% if the CPU. Asterisk then > could be used in hundreds of different roles in the enterprise. PJSIP has > no importance at all, this is the big issue. I suggest the developers look > at an open-source package and adapt the code, is called rtpengine. It uses > a kernel module to do the job. Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200512/0ed9473d/attachment.html>

...packet to detect those which contain DTMF tones, and 'empty' them. First of all, has this already been done? Am I missing some module asterisk already has available that could do this? If not, what would be the best approach: trying to direct RTP through some separate server (eg sipwise/rtpengine) which would implement this, or modifying asterisk to support this? I am really wondering how hard it would be to implement the buffering (holding back of RTP packets for a fraction of a second within asterisk? Alternativey, how hard would it be to configure asterisk so that rtp passed through rtp...

...e from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as Kamailio. The version is 11.10.2. With Kamailio I use rtpengine, which affects SDP descriptions when 488 response is received. My goal is to enable two websocket clients using Chrome to call each other, using Kamailio as outbound proxy. Kamailio routes signaling to Asterisk, and then back to clients. Currently the problem is RTP, when INVITE is received from...

...ebrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine, first one came directly from the client. I defined my clients according to the sip.js guide: http://sipjs.com/guides/server-configuration/asterisk/ So this was rejected: (I marked the extra lines with '//' to ease looking through the differences) v=0 o=- 9046935681162021751 2 IN IP4 91...

dahdi built fine on 5.1.20, but on 5.2.7: ............. CC [M] /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o SHIPPED /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o LD [M] /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_loader.o Building modules, stage 2.