We need to develop a pjsip-based app to bridge calls between WebRTC endpoints and Legacy sip-phones, as below:
[WebRTC endpoint on a browser] <======= webrtc media flow =======> [desired pjsip-based app] <======== sip+rtp flow =======> [SIP/RTP legacy sip-phone]
How can pjsip help with that?
Just my 2 cents... do you really want to develop an app to do this? why
you didn't use opensip as webrtc frontend or asterisk (that use pjsip)
that already do it out-of-the-box?
Il 12/04/23 17:04, jcabezas@neopath.com.br ha scritto:
We need to develop a pjsip-based app to bridge calls between WebRTC
endpoints and Legacy sip-phones, as below:
[WebRTC endpoint on a browser] <======= webrtc media flow =======>
[desired pjsip-based app] <======== sip+rtp flow =======> [SIP/RTP
legacy sip-phone]
How can pjsip help with that?
Visit our blog:http://blog.pjsip.org
pjsip mailing list --pjsip@lists.pjsip.org
To unsubscribe send an email topjsip-leave@lists.pjsip.org
Thanks for replying.
What I am looking for is creating a self-contained component of signalling and media (let’s call it SMLIB) that can be used as part of other, higher-level, components.
As one hypothetical example SMLIB could be used to power a softphone that “talks webrtc”,
Or inside a conference server allowing both legacy SIP and WebRTC-browser clients.
Following your suggestion I should say that OpenSIPS is not a webrtc frontend, it is a medialess (deals only with SIP) component, although in more recent versions it brings WebSockets as a new SIP transport (adding to UDP, TCP, TLS).
Asterisk really supports WebRTC but I suppose I would need to rip-off the webrtc parts of its source code base to use in my Lib, that could be challenging.
Any other thoughts?
Check out SylkServer it does exactly this.
—
Adrian
On 12 Apr 2023, at 12:04, jcabezas@neopath.com.br wrote:
We need to develop a pjsip-based app to bridge calls between WebRTC endpoints and Legacy sip-phones, as below:
[WebRTC endpoint on a browser] <======= webrtc media flow =======> [desired pjsip-based app] <======== sip+rtp flow =======> [SIP/RTP legacy sip-phone]
How can pjsip help with that?
Visit our blog: http://blog.pjsip.org
pjsip mailing list -- pjsip@lists.pjsip.org
To unsubscribe send an email to pjsip-leave@lists.pjsip.org
Thank you for the recommendation.
Downloaded SylkServer source on a Ubuntu server to experiment it.
two questions:
what is the best place to learn the details of APIs, configuration and code integration?
would it be easy to integrate SylkServer in with another running process that would control SylkServer and receive ‘callbacks’ from it in realtime ?
You can look at the Python code and modify it as you see fitted for your integration
https://github.com/AGProjects/sylkserver/tree/master/sylk/applications/webrtcgateway
and use sip beyond voip mailing list for questions.
https://lists.ag-projects.com/mailman/listinfo/sipbeyondvoip
—
Adrian
On 14 Apr 2023, at 12:16, jcabezas@neopath.com.br wrote:
Thank you for the recommendation.
Downloaded SylkServer source on a Ubuntu server to experiment it.
two questions:
what is the best place to learn the details of APIs, configuration and code integration?
would it be easy to integrate SylkServer in with another running process that would control SylkServer and receive ‘callbacks’ from it in realtime ?
Visit our blog: http://blog.pjsip.org
pjsip mailing list -- pjsip@lists.pjsip.org
To unsubscribe send an email to pjsip-leave@lists.pjsip.org