[Asterisk-Users] Asterisk 1.2.8 - WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed

[Erick Perez]

While sending calls to a SIP provider, the following warning generates:

    -- Executing Dial("SIP/1000-c317",
"SIP/13057671523 at 209.120.202.94:5060|55|o") in new stack
    -- Called 13057671523 at 209.120.202.94:5060
    -- SIP/209.120.202.94:5060-0533 is making progress passing it to
SIP/1000-c317
    -- SIP/209.120.202.94:5060-0533 answered SIP/1000-c317
    -- Attempting native bridge of SIP/1000-c317 and
SIP/209.120.202.94:5060-0533
Jun  3 22:56:09 WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker
bit, because SSRC has changed

However the calls complete correctly.
I'm using 1.2.8 asterisk stable release.

what does that mean?

Thanks,


Erick.