[Spce-user] allow source port of RTP to change during a call
Laurent Schweizer
laurent.schweizer at peoplefone.com
Tue Feb 5 11:45:33 EST 2019
Hello,
I'm using the RTPengine with Kamailio and I have a question for a specific case.
I have some customer that are changing the source port of the RTP stream during the call ( no re-invite) I think it's more a NAT issue that a user agent issue...
I see in the doc that I can handle this case with the "media handover" but in that case the rtpengine will allow a change of the source port but also of the source IP.
Any possibility to allow only a change of the source port so source IP must still be the same ?
Thanks
BR
Laurent
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20190205/5f7b3371/attachment.html>
More information about the Spce-user
mailing list