TCP Transport question...

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Hello Benny,

I'm testing pjSIP's behavior using pjsua (latest version) and found out,
that (by my opinion) pjsua has an error when using TCP transport.
I was following instructions from documentation (as below):
------------------------------------------------------------------------
Using TCP Transport

By default, TCP transport will be created and initialized. However, TCP will
not be used automatically unless the destination URL has ";transport=tcp"
parameter in it. (Note: this behavior may change once we support resolving
NAPTR records).

TCP can be specified when registering to server and when sending outgoing
requests. To use TCP when registering, add ";transport=tcp" in the
registrar's URL, for example with "--registrar sip:example.com;transport=tcp"
option.

Similarly ";transport=tcp" parameter needs to be added in the destination
URL when making outgoing calls, subscribing presence, or sending outgoing
MESSAGE request.
------------------------------------------------------------------------

And now, back to the problem:
- First, register via TCP - that went OK.
- Then, in case of an outgoing call, everything is OK too.
- The error occurs at the time when I put this call on hold. The re-INVITE
(sent by pjSIP) that goes out (signaling HOLD condition) is sent via UDP and
not TCP as I would expect!

Am I doing something wrong or is this an error in pjSIP? How can I overcome
this?

Kind regards,
Sasa
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