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 11:42:31.000-0600Date Closed:2011-06-07 14:02:52
Versions:Frequency of
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

Hi Chris,

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

Hi Nir,

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.

Best regards,

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.