[Live-devel] How to handle RTP/RTCP connection to send audio using SIPClient

[Antoine Lavier]

Hi Ross,

Too bad but I understand.

We are currently using the RTSP client part of Live555 and would like to be
able to use the SIP protocol (client/server) as well. Could we consider a
contribution in the future? For the moment, this project is not yet planned
on our side.

Regards,

Antoine Lavier


Le mer. 28 oct. 2020 à 15:59, Ross Finlayson <finlayson at live555.com> a
écrit :

> Bonjour Antoine,
>
> The “playSIP” application was only ever intended to be used to *receive”
> media from a SIP connection.  It cannot be used for two-way communication -
> i.e, to send media as well as receiving it.
>
> Our support for the SIP protocol is very limited (basically, just the
> “playSIP” application, whose code has not been updated in several years).
> Most of our focus currently is on the RTSP protocol, not SIP.
>
> Therefore, I suggest that you try to find some other documentation and/or
> source code to help your project.  Sorry.
>
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
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