|Summary:||ASTERISK-13420: RTP delayed by 30 seconds when SIP calls is bridged via two LOCAL channel.|
|Reporter:||Chris Maciejewski (chris-mac)||Labels:|
|Date Opened:||2009-01-21 11:42:31.000-0600||Date Closed:||2011-06-07 14:02:52|
|Environment:||Attachments:||( 0) extensions.conf|
( 1) full.log
( 2) sip.conf
( 3) test.call
|Description:||My scenario is as follows:|
1a. Call is originated via .call file on a 'Caller' LOCAL channel.
1b. 'Caller' LOCAL channel dials a SIP phone.
2a. When 'Caller' SIP phone answers, it is connected to 'Called' LOCAL channel.
2b. 'Called' LOCAL channel dials another SIP phone.
Problem: RTP stream from 'Called' SIP phone is not relayed to the 'Caller' for the first 30 seconds.
This can be seen in 'full.log' (attached) line 1768 (when Called phone answers) and line 1769 (30 seconds later - when audio is finally delivered to the Caller).
How to reproduce:
1. Edit line 4 of extensions.conf (attached).
2. Copy test.call (attached) to /var/spool/asterisk/outgoing/
3. Answer 'Called' phone.
4. There will be silence for the first 30 seconds.
|Comments:||By: Nir Simionovich (GreenfieldTech - Israel) (greenfieldtech) 2009-01-21 15:09:46.000-0600|
Can you please attach also your sip.conf file, both the [general] section and the corresponding peer contexts?
By: Chris Maciejewski (chris-mac) 2009-01-21 15:12:56.000-0600
Sure, my sip.conf attached.
By: Chris Maciejewski (chris-mac) 2009-01-22 06:16:05.000-0600
Please accept my apology.
This "bug" turned out to be miss configured firewall on my development machine.
After correcting iptables rules everything works as expected. No delay in RTP streams.
By: Michiel van Baak (mvanbaak) 2009-01-22 06:20:27.000-0600
Thanks for reporting back.
Closing issue as it's a configuration error.