[asterisk-dev] Why does chan_sip disable early bridging???

[Power, Paul C.]

I got this working on 1.2.9 and 1.4.11.

We use Polycom phones with and OpenSER proxy and AudioCodes gateway to
the PSTN world.
Progressinband is set to never.

 

> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com 
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of 
> Chris Ziomkowski
> Sent: Wednesday, October 03, 2007 1:27 AM
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] Why does chan_sip disable early 
> bridging???
> 
> Not sure how that would work, given that the code below 
> specifically prohibits it. What version are you running? This 
> is the latest code from 1.4.11. I have never used the "r" 
> option in Dial as I want to hear remote ringback. I never 
> want to genererate it locally.
> 
> For some reason, SIP channels are specifically prohibited for 
> early media except in the case where you have specified 
> Direct RTP according to the code below. (which is confusing 
> in an of itself because if you are using direct rtp then you 
> don't need Asterisk to do media routing, early or otherwise, do you?)
> 
> In any case, the line in chan_sip.c appears to be incorrect 
> and I have corrected it in my version. I would hope the 
> maintainers can look into this and either explain what the 
> original purpose of the statement was, or else remove it if 
> it is in fact a bug.
> 
> 
> Power, Paul C. wrote:
> 
> >I got early media to work for me by omitting the 'r' option 
> in the dial 
> >statement.
> >
> >  
> >
> >>-----Original Message-----
> >>From: asterisk-dev-bounces at lists.digium.com
> >>[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Chris 
> >>Ziomkowski
> >>Sent: Monday, October 01, 2007 4:58 AM
> >>To: asterisk-dev at lists.digium.com
> >>Subject: [asterisk-dev] Why does chan_sip disable early bridging???
> >>
> >>OK everyone,
> >>
> >>I have been banging my head against a wall for 2 days 
> trying to figure 
> >>out why I can't get ringback on my SIP to SIP calls through 
> Asterisk.
> >>Eventually, I tracked it down. The call to ast_early_bridge 
> actually 
> >>completes without error, but the problem appears in chan_sip, in
> >>sip_set_rtp_peer() at or about line 17204 in the 1.4.11 version of 
> >>Asterisk:
> >>
> >>if (chan->_state != AST_STATE_UP && !global_directrtpsetup) 
>     /* We 
> >>are in early state */
> >>    return 0;
> >> 
> >>Apparently, any attempt at early media over SIP is thwarted in the 
> >>chan_sip module.  After commenting out  this line, my calls bridge 
> >>fine and I hear the ringback and other progress messages. I 
> have not 
> >>yet noticed any ill effects from doing this.
> >>
> >>My question is: what was the purpose of this statement? Is there 
> >>another/better way to get early media on SIP to SIP calls?
> >>
> >>Thanks in advance,
> >>
> >>Chris
> >>
> >>
> >>
> >>
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> >
> >
> >  
> >
> 
> 
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