Hi,
we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One
GrandStream GXV3000 is used for the tests. He is registered to asterisk
1.6.2.18 asa well as 1.4.35. Calling echo test is OK on both servers,
get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP
trunk from both others servers is also working well.
What fail, is video on echo test from asterisk 1.4.42 using SIP trunks:
we have audio but no videobeside the fact that video codec are
negociated as shown below.
All servers are on public IP. Here is a debug from a call from server
running 1.4.35 asterisk to the 1.4.42 one:
<------------->
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: --- (14 headers 18 lines)
---
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Sending to
XXX.XXX.XXX.XXX : 5060 (no NAT)
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Using INVITE request as
basis request - 78938c042d374b341c4f1b60071d3...@xxx.xxx.xxx.xxx
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found peer 'mypeer'
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 0
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 3
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 101
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description
format PCMU for ID 0
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description
format GSM for ID 3
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description
format telephone-event for ID 101
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 34
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 103
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 99
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description
format H263 for ID 34
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description
format h263-1998 for ID 103
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description
format H264 for ID 99
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Capabilities: us -
0x3c0002 (gsm|h261|h263|h263p|h264), peer - audio=0x380006
(gsm|ulaw|h263|h263p|h264)/video=0x380000 (h263|h263p|h264), combined -
0x380002 (gsm|h263|h263p|h264)
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Non-codec capabilities
(dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event),
combined - 0x1 (telephone-event)
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Peer audio RTP is at port
XXX.XXX.XXX.XXX:40428
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Peer video RTP is at port
XXX.XXX.XXX.XXX:44636
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Looking for 3800 in
acces_groupe (domain mydomain.com)
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: list_route: hop:
<sip:3...@xxx.xxx.xxx.xxx>
[2011-07-05 16:08:14] VERBOSE[11535] logger.c:
<--- Transmitting (NAT) to XXX.XXX.XXX.XXX:5060 --->
All trunks. are setted from the same manier:
[trunk]
;
type=peer
deny=0.0.0.0/0.0.0.0
permit=XXX.XXX.XXX.XXX
host=host.domain.com
context=from-trunk
disallow=all
allow=all
A "sip show peer <mypeer>" show that video is on.
What can be the problem, I start loose my hairs!
Thanks for any hint.
--
Daniel
--
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