Summary:ASTERISK-01964: [patch] Rtp media stream doesn't honor bindaddr if the addr is an alias on an interface
Reporter:Matteo Brancaleoni (mbrancaleoni)Labels:
Date Opened:2004-07-07 14:43:09Date Closed:2008-01-15 15:01:52.000-0600
Versions:Frequency of
Environment:Attachments:( 0) rtp.patch
Description:If bindaddr into sip.conf is on a ip alias (eg. eth0 is and the alias eth0:1 is, the signalling works correctly on (eth0:1) but the rtp flows from eth0, aka

I think that can be simply fixed in ast_rtp_new by passing also the ipaddr and so setting rtp->us.sin_addr to the ip.
if the ip is NULL, just work as usual.

or is better to add a bindaddr option also into rtp.conf?

I can write that small patch, just lemme know what could be the best way to handle that.
Comments:By: Matteo Brancaleoni (mbrancaleoni) 2004-07-07 15:25:55

this simple patch add the param bindaddr=x.y.z.k into rtp.conf, in order to make rtp binding to a specific ip if needed.

please check it out. was written in 5 minutes, although seems to work here.

not specifying the bindaddr= param makes rtp work as usual.

By: Brian West (bkw918) 2004-07-07 16:25:48

we have now maybe 4 of these patch on the bug tracker... all diffrent.  Talk to kram so you can do it properly.

By: Mark Spencer (markster) 2004-07-08 08:00:42

Okay update to latest CVS and see if that does it for you.  This method just has SIP use its bindaddress for calling the RTP.

By: Mark Spencer (markster) 2004-07-09 05:15:25

I'm going to assume this fixed, you can reopen if it isn't.

By: Digium Subversion (svnbot) 2008-01-15 15:01:52.000-0600

Repository: asterisk
Revision: 3393

U   trunk/channels/chan_sip.c
U   trunk/include/asterisk/rtp.h
U   trunk/rtp.c

r3393 | markster | 2008-01-15 15:01:52 -0600 (Tue, 15 Jan 2008) | 2 lines

Extend bindaddr to RTP connections on SIP (bug ASTERISK-1964 et al)