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

Chris Ziomkowski cziom at jsg.co.th
Wed Oct 3 02:26:52 CDT 2007


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. 
>
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>>-----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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