[asterisk-users] video calls not working
Danny Nicholas
danny at debsinc.com
Mon Nov 21 09:07:47 CST 2011
Two items
#1 you only need 1 disallow=all in your sip.conf definition
#2 you need to patch rtp.c to define 126 as FORMAT_H263 - this is an xlite
response to Asterisk starting music-on-hold during the connect pause. The r
on the dial command attempts to do a "faux ring" which xlite interprets as a
MOH request, so if you don't want to patch/recompile, just take the r off of
Dial.
From: virendra bhati [mailto:virbhati at gmail.com]
Sent: Monday, November 21, 2011 4:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Sam Govind;
Danny Nicholas
Subject: video calls not working
Hi list,
I am not able to make video calls between two sip accounts. below is the
information. please help me where I am missing the configuration.
Extensions.conf
exten => 111,1,Answer()
same => n,Dial(SIP/2206,60,r)
same => n,Hangup()
SIP.conf
[2218]
type=friend
secret=*******
callerid="Virendra" <9172341457>
host=dynamic ; This device needs to register
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
dtmfmode=inband
insecure=invite,port
;context=outbound
context=bhati-test
qualify=yes
accountcode=123654789
disallow = all
allow = ulaw,alaw,h263,g729,gsm,h264
videosupport=yes
[2206]
type=friend
secret=*******
callerid=2206
host=dynamic ; This device needs to register
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
dtmfmode=inband
insecure=invite,port
context=outbound
qualify=yes
disallow = all
allow = ulaw,alaw,h263,g729,gsm,h264
videosupport=yes
codec list of asterisk 1.6.2.11
haddock8-astrx*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME DESC
----------------------------------------------------------------------------
----
1 (1 << 0) (0x1) audio g723 (G.723.1)
2 (1 << 1) (0x2) audio gsm (GSM)
4 (1 << 2) (0x4) audio ulaw (G.711 u-law)
8 (1 << 3) (0x8) audio alaw (G.711 A-law)
16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2)
32 (1 << 5) (0x20) audio adpcm (ADPCM)
64 (1 << 6) (0x40) audio slin (16 bit Signed Linear
PCM)
128 (1 << 7) (0x80) audio lpc10 (LPC10)
256 (1 << 8) (0x100) audio g729 (G.729A)
512 (1 << 9) (0x200) audio speex (SpeeX)
1024 (1 << 10) (0x400) audio ilbc (iLBC)
2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551)
4096 (1 << 12) (0x1000) audio g722 (G722)
65536 (1 << 16) (0x10000) image jpeg (JPEG image)
131072 (1 << 17) (0x20000) image png (PNG image)
262144 (1 << 18) (0x40000) video h261 (H.261 Video)
524288 (1 << 19) (0x80000) video h263 (H.263 Video)
1048576 (1 << 20) (0x100000) video h263p (H.263+ Video)
2097152 (1 << 21) (0x200000) video h264 (H.264 Video)
haddock8-astrx*CLI>
CLI Output:-
-- Executing [111 at bhati-test:1] Answer("SIP/2218-00000664", "") in new
stack
-- Executing [111 at bhati-test:2] Dial("SIP/2218-00000664",
"SIP/2206,60,r") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Called 2206
-- SIP/2206-00000665 is ringing
-- SIP/2206-00000665 is ringing
[Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
-- SIP/2206-00000665 answered SIP/2218-00000664
[Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:57:30] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:57:34] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:57:40] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:57:44] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:57:50] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:57:54] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:58:00] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:58:04] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:58:11] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:58:13] NOTICE[7924]: chan_sip.c:21479 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 2206
[Nov 21 15:58:15] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:58:21] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:58:25] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:58:30] NOTICE[7924]: chan_sip.c:21479 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 2218
[Nov 21 15:58:31] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:58:35] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:58:41] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
-- Executing [h at bhati-test:1] NoOp("SIP/2218-00000664", "hangup the call
now") in new stack
== Spawn extension (bhati-test, 111, 2) exited non-zero on
'SIP/2218-00000664'
--
-----
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
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