[sip-comm-dev] Possible Issue with SDP & RTP


#1

Hi,

Javier Mendiara Ca�ardo schrieb:

As you can see, the other softphone is sending RTP directly to SipComm,
but SipComm is sending audio via SIP Server.

I think this is related to SIP REINVITES, as they are (afaik) currently
not supported by sc.
If you have a SIP client that does not support REINVITES, the audio is
send via the SIP server, only if REINVITES are supported, this data is
send directly between the two clients.

Still I think that SC's handling is somewhere wrong, as I have the same
issue. But normally one should not care whether the client supports
REINVITES or not, this should be transparently to the user.

Btw: we fixed this here in that way, that we set the "support reinvites
flag" on the asterisk server to false for those accounts using sc as
their SIP client.

Sebastian

···

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