Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Auto video call hangup


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
mvakondios at gmail.com
Guest





PostPosted: Thu Oct 23, 2014 9:58 am    Post subject: [asterisk-users] Auto video call hangup Reply with quote

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@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@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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services