[Live-devel] How to get other endpoint's IP address

[Ben Rush]

Hi Ross,

Thanks for you feedback again. One question. I'm confused about the order
of operations, then. You claim I should call .changeDestinationparameters()
with the sender's IP on the rtcpGroupSock before invoking
RTCPInstance::createNew() on said socket, but before receiving the sender's
IP I need to get traffic from it, and before getting traffic, don't I need
to have invoked RTCPInstance::createNew()?

My understanding of the order is:

1. Create sink,
2. Create sockets,
3. Create RTP source,
4. Create RTCP instance,
5. Receive audio traffic, get sender's IP.
6. Then maybe update stuff with changeDestinationParameters()?

Or am I misunderstanding what you're saying? Apologies if I'm being dense.

On Tue, Jun 14, 2016 at 1:39 AM Ross Finlayson <finlayson at live555.com>
wrote:

> > I'm creating an audio "client" using RTPSource. I have a custom
> MediaSink-derived class which takes the traffic and routes it to my
> computer's audio device. Works great.
> >
> > What I'd like to do, however, is retrieve the sender's IP address (the
> IP address of the sender which is sending traffic to the port/ip as
> specified by my GroupSocks). How do I do this?
>
> Unfortunately the sender’s IP address (and port number) aren’t exposed
> directly by our “*RTPSource” code, but there is a hack that you can use to
> get this.
>
> The trick (hack) is to define a subclass of the “Groupsock” class, and, in
> your subclass, reimplement the virtual function
>         virtual Boolean handleRead(unsigned char* buffer, unsigned
> bufferMaxSize, unsigned& bytesRead, struct sockaddr_in& fromAddressAndPort);
> In your subclass, you would implement this virtual function as follows:
>         Boolean yourGroupsockSubclass::handleRead(unsigned char* buffer,
> unsigned bufferMaxSize,
>                                                 unsigned& bytesRead,
> struct sockaddr_in& fromAddressAndPort) {
>                 Boolean result = Groupsock::handleRead(buffer,
> bufferMaxSize, bytesRead, fromAddressAndPort);
>                 if (result) {
>                         // “fromAddressAndPort” is the sender’s IP address
> (and port number); record it
>                 }
>
>                 return result;
>         }
>
> Then, you would create your “RTPSource” object with an object of your
> “Groupsock" subclass, rather than with a regular “Groupsock”.
>
> Also:
> > RTCPInstance* rtcpInstance =
> >         RTCPInstance::createNew(*environment, &rtcpGroupSock, 5000,
> CNAME, NULL, rtpSource);
>
> Before doing this, you should call
>         rtcpGroupSock.changeDestinationParameters()
> with the sender’s IP address and port number, so that RTCP “RR” packets go
> back to the sender.  (And if you want to use “rtpGroupSock” to send audio
> back to the sender (via a “RTPSink” object), then you should call
> “changeDestinationParameters()” on “rtpGroupSock” as well.)
>
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
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> live-devel mailing list
> live-devel at lists.live555.com
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>
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