[asterisk-users] Auto video call hangup

Markos Vakondios mvakondios at gmail.com
Thu Oct 23 09:57:56 CDT 2014


Hi,

I use a simple scheme:

SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video
phone B (h264/Asterisk 11.7.0)

When calls from A to B and vice versa drop on pickup.

On B side:

[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the
marker bit due to a source update
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the
marker bit due to a source update
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the
marker bit due to a source update
[Oct 24 16:33:49] WARNING[15202] chan_iax2.c: Received mini frame before
first full video frame
[Oct 24 16:33:49] DEBUG[15206] chan_iax2.c: Ooh, video format changed to
h264
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Ooh, format
changed from unknown to h264
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Starting
RTCP transmission on RTP instance '0xb7d79c54'
[Oct 24 16:33:49] DEBUG[15207] chan_iax2.c: Ooh, voice format changed to
'ulaw'
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Ooh, format
changed from unknown to ulaw
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Created
smoother: format: ulaw ms: 20 len: 160
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Starting
RTCP transmission on RTP instance '0xb69b9894'
[Oct 24 16:33:49] DEBUG[15204] chan_iax2.c: Packet arrived out of order
(expecting 7, got 5) (frametype = 3, subclass = 200004)
[Oct 24 16:33:49] DEBUG[15204] chan_iax2.c: Acking anyway
[Oct 24 16:33:49] DEBUG[15211] chan_iax2.c: Packet arrived out of order
(expecting 7, got 6) (frametype = 2, subclass = 100003)
[Oct 24 16:33:49] DEBUG[15211] chan_iax2.c: Acking anyway
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Difference
is 45450, ms is 505 (45450), pred/ts/samples 45450/0/0
[Oct 24 16:33:50] DEBUG[15193][C-00000012] chan_sip.c: Trying to put
'SIP/2.0 200' onto UDP socket destined for 192.168.0.192:5060
[Oct 24 16:33:50] DEBUG[15590][C-00000012] res_rtp_asterisk.c: 0xb7d93ce0
-- Probation learning mode pass with source address 192.168.0.192:5004
[Oct 24 16:33:50] DEBUG[15590][C-00000012] res_rtp_asterisk.c: 0xb69b5488
-- Probation learning mode pass with source address 192.168.0.192:5006
[Oct 24 16:33:50] WARNING[15590][C-00000012] chan_iax2.c: Can't compress
subclass 2097217
[Oct 24 16:33:50] DEBUG[15207] chan_iax2.c: Received VNAK: resending
outstanding frames
[Oct 24 16:33:50] DEBUG[15209] chan_iax2.c: Received VNAK: resending
outstanding frames
[Oct 24 16:33:50] DEBUG[15208] chan_iax2.c: Received VNAK: resending
outstanding frames
[Oct 24 16:33:50] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the
marker bit due to a source update
[Oct 24 16:33:50] DEBUG[15204] chan_iax2.c: Received VNAK: resending
outstanding frames
[Oct 24 16:33:50] DEBUG[15211] chan_iax2.c: Received VNAK: resending
outstanding frames
[Oct 24 16:33:50] DEBUG[15203] chan_iax2.c: Received VNAK: resending
outstanding frames
[Oct 24 16:33:50] DEBUG[15202] chan_iax2.c: Received VNAK: resending
outstanding frames
[Oct 24 16:33:50] DEBUG[15206] chan_iax2.c: Received VNAK: resending
outstanding frames
[Oct 24 16:33:50] DEBUG[15205] chan_iax2.c: Immediately destroying 1664,
having received hangup
[Oct 24 16:33:50] DEBUG[15227] manager.c: Examining event:
[Oct 24 16:33:50] DEBUG[15590][C-00000012] channel.c: Didn't get a frame
from channel: IAX2/THNS-1664
[Oct 24 16:33:50] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the
marker bit due to a source update
[Oct 24 16:33:50] DEBUG[15590][C-00000012] channel.c: Bridge stops bridging
channels IAX2/THNS-1664 and SIP/507-0000001d
[Oct 24 16:33:50] DEBUG[15590][C-00000012] channel.c: Soft-Hanging up
channel 'IAX2/THNS-1664'
[Oct 24 16:33:50] DEBUG[15590][C-00000012] pbx.c: Launching 'Macro'
[Oct 24 16:33:50] VERBOSE[15590][C-00000012] pbx.c:     -- Executing
[h at macro-dial-one:1] Macro("IAX2/THNS-1664", "hangupcall,") in new stack

Debug at A side:

[Oct 23 17:42:47] WARNING[14880] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] WARNING[14887] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] WARNING[14881] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] WARNING[14885] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] WARNING[22489] res_rtp_asterisk.c: Don't know how to send
format unknown packets with RTP
[Oct 23 17:42:47] WARNING[14886] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] WARNING[14879] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] WARNING[14878] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] WARNING[14880] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] WARNING[14887] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] WARNING[14881] chan_iax2.c: Received mini frame before
first full video frame
[Oct 23 17:42:47] VERBOSE[22489] pbx.c:     -- Executing
[h at macro-dialout-trunk:1] Macro("SIP/102-00000098", "hangupcall,") in new
stack



I have also tested with the following setup and video is displayed
correctly (the only difference is the asterisk version of B)

SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video
phone B (h264/Asterisk 1.8.20)

Any pointers to help me debug further please? Does had a similar problem?
The videophones used A:Grandstream GXV-3000/3140 B:Grandstream GXV-3275

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141023/65794b93/attachment.html>


More information about the asterisk-users mailing list