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Did anyone manage to use SipUnit for testing RTP? (e.g. sending audio/video from one SipPhone to another and validate the correct content). My understanding is that this is not currently supported by SipUnit; maybe it is possible with the help of another library ... any suggestion is appreciated.
Thank you!
The text was updated successfully, but these errors were encountered:
Hello,
Great work with this framework!
Did anyone manage to use SipUnit for testing RTP? (e.g. sending audio/video from one SipPhone to another and validate the correct content). My understanding is that this is not currently supported by SipUnit; maybe it is possible with the help of another library ... any suggestion is appreciated.
Thank you!
The text was updated successfully, but these errors were encountered: