Posted: Wed Jul 05, 2006 11:11 am Post subject: [Asterisk-video] Asterisk H264
So do you actually see h264 when you run a 'show codec'? I can't
seem to get that...though i do have the file on my modules
directory.... Oh..and it isn't working with my endpoint either......
am i missing a dependency or something?
So do you actually see h264 when you run a 'show codec'? I can't
seem to get that...though i do have the file on my modules
directory.... Oh..and it isn't working with my endpoint either......
am i missing a dependency or something?
--- (13 headers 21 lines)---
Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
Using INVITE request as basis request -
9c00870094007300@xxx.xxx.xxx.xxx
Found user '3005'
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP video format 31
Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
Found description format G7221 for ID 96
Found description format G7221 for ID 97
Found description format G722 for ID 9
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format H261 for ID 31
Capabilities: us - 0x28070f (g723|gsm|ulaw|alaw|g729|speex|ilbc|
h263|h264), peer - audio=0x4040c (ulaw|alaw|ilbc|h261)/
video=0x40000 (h261), combined - 0x40c (ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
Peer video RTP is at port xxx.xxx.xxx.xxx:5602
Looking for 3001 in from-sip (domain aaa.aaa.aaa.aaa)
list_route: hop: <sip:3005@xxx.xxx.xxx.xxx>
--- (13 headers 21 lines)---
Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
Using INVITE request as basis request -
9c00870094007300@xxx.xxx.xxx.xxx
Found user '3005'
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP video format 31
Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
Found description format G7221 for ID 96
Found description format G7221 for ID 97
Found description format G722 for ID 9
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format H261 for ID 31
Capabilities: us - 0x28070f (g723|gsm|ulaw|alaw|g729|speex|ilbc|
h263|h264), peer - audio=0x4040c (ulaw|alaw|ilbc|h261)/
video=0x40000 (h261), combined - 0x40c (ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
Peer video RTP is at port xxx.xxx.xxx.xxx:5602
Looking for 3001 in from-sip (domain aaa.aaa.aaa.aaa)
list_route: hop: <sip:3005@xxx.xxx.xxx.xxx>
Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:
5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x
xx.xxx.xxx.xxx
From: "BK" <sip:
3005@aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
To: <sip:3001@aaa.aaa.aaa.aaa>
Call-ID: 9c00870094007300@xxx.xxx.xxx.xxx
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:3001@aaa.aaa.aaa.aaa>
Content-Length: 0
---
-- Executing [3001@from-sip:1] Dial("SIP/3005-08dd1948", "SIP/
3001|20") in new stack
-- Executing [3001@from-sip:1] Dial("SIP/3005-08dd1948", "SIP/
3001|20") in new stack
Video is at aaa.aaa.aaa.aaa port 11862
Audio is at aaa.aaa.aaa.aaa port 17218
Video is at aaa.aaa.aaa.aaa port 11862
Audio is at aaa.aaa.aaa.aaa port 17218
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
INVITE sip:yyy.yyy.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK0fcd45d2;rport
From: "BK" <sip:3005@aaa.aaa.aaa.aaa>;tag=as0294a8fa
To: <sip:yyy.yyy.yyy.yyy>
Contact: <sip:3005@aaa.aaa.aaa.aaa>
Call-ID: 4810f285311c63fc642bc5b86fd3f168@aaa.aaa.aaa.aaa
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 05 Jul 2006 21:12:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 274
--- (11 headers 12 lines)---
Found RTP audio format 8
Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
Found description format PCMA for ID 8
Capabilities: us - 0x380408 (alaw|ilbc|h263|h263p|h264), peer -
audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
list_route: hop: <sip:yyy.yyy.yyy.yyy>
set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to
send to
set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to
send to
nyguest25*CLI>
---
set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to
send to
set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
BYE sip:yyy.yyy.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK64b8feae;rport
From: "BK" <sip:3005@aaa.aaa.aaa.aaa>;tag=as0294a8fa
To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
Contact: <sip:3005@aaa.aaa.aaa.aaa>
Call-ID: 4810f285311c63fc642bc5b86fd3f168@aaa.aaa.aaa.aaa
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (from-sip, 3001, 1) exited non-zero on 'SIP/
3005-08dd1948'
set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
BYE sip:yyy.yyy.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK64b8feae;rport
From: "BK" <sip:3005@aaa.aaa.aaa.aaa>;tag=as0294a8fa
To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
Contact: <sip:3005@aaa.aaa.aaa.aaa>
Call-ID: 4810f285311c63fc642bc5b86fd3f168@aaa.aaa.aaa.aaa
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
> "show codecs" has a "bug" in it (even in trunk) where it doesn't
> go far
> enough to display the H.264 codec.
>
> What version of Asterisk are you using?
>
> What's the endpoint you're using - some don't like the way that
> Asterisk
> currently don't send fmtp in the SDP.
>
> John Martin
> http://www.AuPix.com
>
Hmmmm.... I can't see either H.263 or H.264 in the Tandberg sdp, there's
only H.261!
Can you ethereal it to see if it's in the SDP on the wire?
John Martin
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Chaim
Fried
Sent: 05 July 2006 22:38
To: Chaim Fried
Cc: Development discussion of video media support in Asterisk
Subject: Re: [Asterisk-video] Asterisk H264
On Jul 5, 2006, at 5:24 PM, Chaim Fried wrote:
Quote:
i am on the latest trunk....
i am using tandberg video endpoints. they work fine with Asterisk
on h.263 but not h.264..
here is a sip debug snippet...
---
Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
Using INVITE request as basis request -
9c00870094007300@xxx.xxx.xxx.xxx
Scheduling destruction of SIP dialog
'9c00870094007300@xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE)
Found user '3005'
Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:
5060;branch=z9hG4bKa4c54c13aa2c5f75f3b1a611eaa51cd8.1;rport;received=x
--- (13 headers 21 lines)---
Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
Using INVITE request as basis request -
9c00870094007300@xxx.xxx.xxx.xxx
Found user '3005'
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP video format 31
Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
Found description format G7221 for ID 96
Found description format G7221 for ID 97
Found description format G722 for ID 9
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format H261 for ID 31
Capabilities: us - 0x28070f (g723|gsm|ulaw|alaw|g729|speex|ilbc|
h263|h264), peer - audio=0x4040c (ulaw|alaw|ilbc|h261)/
video=0x40000 (h261), combined - 0x40c (ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
Peer video RTP is at port xxx.xxx.xxx.xxx:5602
Looking for 3001 in from-sip (domain aaa.aaa.aaa.aaa)
list_route: hop: <sip:3005@xxx.xxx.xxx.xxx>
--- (13 headers 21 lines)---
Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
Using INVITE request as basis request -
9c00870094007300@xxx.xxx.xxx.xxx
Found user '3005'
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP video format 31
Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
Found description format G7221 for ID 96
Found description format G7221 for ID 97
Found description format G722 for ID 9
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format H261 for ID 31
Capabilities: us - 0x28070f (g723|gsm|ulaw|alaw|g729|speex|ilbc|
h263|h264), peer - audio=0x4040c (ulaw|alaw|ilbc|h261)/
video=0x40000 (h261), combined - 0x40c (ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
Peer video RTP is at port xxx.xxx.xxx.xxx:5602
Looking for 3001 in from-sip (domain aaa.aaa.aaa.aaa)
list_route: hop: <sip:3005@xxx.xxx.xxx.xxx>
Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:
5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x
---
-- Executing [3001@from-sip:1] Dial("SIP/3005-08dd1948", "SIP/
3001|20") in new stack
-- Executing [3001@from-sip:1] Dial("SIP/3005-08dd1948", "SIP/
3001|20") in new stack
Video is at aaa.aaa.aaa.aaa port 11862
Audio is at aaa.aaa.aaa.aaa port 17218
Video is at aaa.aaa.aaa.aaa port 11862
Audio is at aaa.aaa.aaa.aaa port 17218
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
INVITE sip:yyy.yyy.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK0fcd45d2;rport
From: "BK" <sip:3005@aaa.aaa.aaa.aaa>;tag=as0294a8fa
To: <sip:yyy.yyy.yyy.yyy>
Contact: <sip:3005@aaa.aaa.aaa.aaa>
Call-ID: 4810f285311c63fc642bc5b86fd3f168@aaa.aaa.aaa.aaa
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 05 Jul 2006 21:12:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 274
--- (11 headers 12 lines)---
Found RTP audio format 8
Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
Found description format PCMA for ID 8
Capabilities: us - 0x380408 (alaw|ilbc|h263|h263p|h264), peer -
audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
list_route: hop: <sip:yyy.yyy.yyy.yyy>
set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to
send to
v=0
o=tandberg 0 1 IN IP4 yyy.yyy.yyy.yyy
s=-
set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
IP4 yyy.yyy.yyy.yyy
b=CT:768
t=0 0
m=audio 5600 RTP/AVP 8
c=IN IP4 yyy.yyy.yyy.yyy
b=TIAS:64000
a=sendrecv
a=rtpmap:8 PCMA/8000
a=maxprate:50.0
--- (11 headers 12 lines)---
Found RTP audio format 8
Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
Found description format PCMA for ID 8
Capabilities: us - 0x380408 (alaw|ilbc|h263|h263p|h264), peer -
audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:ephone-event), peer -
0x0 (nothing), combined -
ACK sip:yyy.yyy.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK60a3345b;rport
From: "BK" <sip:3005@aaa.aaa.aaa.aaa>;tag=as0294a8fa
To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
Contact: <sip:3005@aaa.aaa.aaa.aaa>
Call-ID: 4810f285311c63fc642bc5b86fd3f168@aaa.aaa.aaa.aaa
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/3001-08dd9bf8 answered SIP/3005-08dd1948
0x0 (nothing)
Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
list_route: hop: <sip:yyy.yyy.yyy.yyy>
set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to
send to
set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
ACK sip:yyy.yyy.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK60a3345b;rport
From: "BK" <sip:3005@aaa.aaa.aaa.aaa>;tag=as0294a8fa
To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
Contact: <sip:3005@aaa.aaa.aaa.aaa>
Call-ID: 4810f285311c63fc642bc5b86fd3f168@aaa.aaa.aaa.aaa
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/3001-08dd9bf8 answered SIP/3005-08dd1948
Audio is at aaa.aaa.aaa.aaa port 15984
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:
5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x
---
-- Native bridging SIP/3005-08dd1948 and SIP/3001-08dd9bf8
Audio is at aaa.aaa.aaa.aaa port 15984
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:
5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x
set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to
send to
nyguest25*CLI>
---
set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to
send to
set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
BYE sip:yyy.yyy.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK64b8feae;rport
From: "BK" <sip:3005@aaa.aaa.aaa.aaa>;tag=as0294a8fa
To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
Contact: <sip:3005@aaa.aaa.aaa.aaa>
Call-ID: 4810f285311c63fc642bc5b86fd3f168@aaa.aaa.aaa.aaa
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (from-sip, 3001, 1) exited non-zero on 'SIP/
3005-08dd1948'
set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
BYE sip:yyy.yyy.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK64b8feae;rport
From: "BK" <sip:3005@aaa.aaa.aaa.aaa>;tag=as0294a8fa
To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
Contact: <sip:3005@aaa.aaa.aaa.aaa>
Call-ID: 4810f285311c63fc642bc5b86fd3f168@aaa.aaa.aaa.aaa
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
> "show codecs" has a "bug" in it (even in trunk) where it doesn't
> go far
> enough to display the H.264 codec.
>
> What version of Asterisk are you using?
>
> What's the endpoint you're using - some don't like the way that
> Asterisk
> currently don't send fmtp in the SDP.
>
> John Martin
> http://www.AuPix.com
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
So do you actually see h264 when you run a 'show codec'? I can't
seem to get that...though i do have the file on my modules directory....
Oh..and it isn't working with my endpoint either......
am i missing a dependency or something?
Posted: Tue Jul 11, 2006 5:45 am Post subject: [Asterisk-video] Asterisk H264
Hi All,
Can we use GXV3000 (h264) based phones with current trunk version?
I tried this couple of weeks ago and it was not working.
Kind regards,
Ahsan
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Trond G.
Andersen
Sent: 06 July 2006 05:47
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] Asterisk H264
Are you using a tandberg 1000 or 150 ? Which SW version are you using.
I know there has been some problems with H264 and Asterisk in older
versions..
So do you actually see h264 when you run a 'show codec'? I can't
seem to get that...though i do have the file on my modules directory....
Oh..and it isn't working with my endpoint either......
am i missing a dependency or something?
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Ahsan
Masood
Sent: 11 July 2006 15:52
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] Asterisk H264
Hi All,
Can we use GXV3000 (h264) based phones with current trunk version?
I tried this couple of weeks ago and it was not working.
Kind regards,
Ahsan
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Trond G.
Andersen
Sent: 06 July 2006 05:47
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] Asterisk H264
Are you using a tandberg 1000 or 150 ? Which SW version are you using.
I know there has been some problems with H264 and Asterisk in older
versions..
So do you actually see h264 when you run a 'show codec'? I can't
seem to get that...though i do have the file on my modules directory....
Oh..and it isn't working with my endpoint either......
am i missing a dependency or something?
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Posted: Tue Jul 18, 2006 10:32 am Post subject: [Asterisk-video] Asterisk H264
Could you please please please let me know what's needed to pull this off? I
do have the exact same setup and am trying to feed some video from Asterisk
directly to my GXV-3000, thus far I have no clue as to how to do this.
Anyway, I checked out Asterisk via subversion two days ago - is that the
trunk? (saw someone talk about 1.4 somewhere).
Your help would be greatly appreciated ;-)
Best,
Michael
----- Original Message -----
From: "Ahsan Masood" <ahsan@telappliant.com>
To: "Development discussion of video media support in Asterisk"
<asterisk-video@lists.digium.com>
Sent: Tuesday, July 18, 2006 11:43 AM
Subject: RE: [Asterisk-video] Asterisk H264
Hi,
I have installed the trunk version today and I can see H264 in show
codecs.
2097152 (1 << 21) (0x200000) video h264 (H.264 Video)
I am using two GXV-3000 phones, there is no video in the call just
audio.
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Ahsan
Masood
Sent: 11 July 2006 15:52
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] Asterisk H264
Hi All,
Can we use GXV3000 (h264) based phones with current trunk version?
I tried this couple of weeks ago and it was not working.
Kind regards,
Ahsan
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Trond G.
Andersen
Sent: 06 July 2006 05:47
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] Asterisk H264
Are you using a tandberg 1000 or 150 ? Which SW version are you using.
I know there has been some problems with H264 and Asterisk in older
versions..
So do you actually see h264 when you run a 'show codec'? I can't
seem to get that...though i do have the file on my modules directory....
Oh..and it isn't working with my endpoint either......
am i missing a dependency or something?
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Posted: Tue Jul 18, 2006 11:04 am Post subject: [Asterisk-video] Asterisk H264
I configured the 'allow=h264' for my two lines and Asterisk complains on
reload:
'cannot allow unkown format 'h264'
Does this happen to anyone else? Please type 'reload' on the asterisk
command line and let me know.
Michael
----- Original Message -----
From: "Ahsan Masood" <ahsan@telappliant.com>
To: "Development discussion of video media support in Asterisk"
<asterisk-video@lists.digium.com>
Sent: Tuesday, July 18, 2006 11:43 AM
Subject: RE: [Asterisk-video] Asterisk H264
Hi,
I have installed the trunk version today and I can see H264 in show
codecs.
2097152 (1 << 21) (0x200000) video h264 (H.264 Video)
I am using two GXV-3000 phones, there is no video in the call just
audio.
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Ahsan
Masood
Sent: 11 July 2006 15:52
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] Asterisk H264
Hi All,
Can we use GXV3000 (h264) based phones with current trunk version?
I tried this couple of weeks ago and it was not working.
Kind regards,
Ahsan
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Trond G.
Andersen
Sent: 06 July 2006 05:47
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] Asterisk H264
Are you using a tandberg 1000 or 150 ? Which SW version are you using.
I know there has been some problems with H264 and Asterisk in older
versions..
So do you actually see h264 when you run a 'show codec'? I can't
seem to get that...though i do have the file on my modules directory....
Oh..and it isn't working with my endpoint either......
am i missing a dependency or something?
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
_______________________________________________
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Posted: Wed Jul 19, 2006 4:09 am Post subject: [Asterisk-video] Asterisk H264
I think that I've spotted where the issue is with asterisk
in middle of 2 h264 streams.
H264 requires in SDP the format of the codec. When you make
a call with H264 there's a a:ftmp that specifies at least
the profile-level-id of the h264 stream.
Asterisk should copy that information to the other peer,
in addition to the rtpmap.
Posted: Wed Jul 19, 2006 4:35 am Post subject: [Asterisk-video] Asterisk H264
Hi Matteo,
Some H.264 endpoints do indeed need the fmtp to be specified before
they will make a video call. Some just assume defaults - though it can
mean you end up with a low resolution, low frame rate call. Some
endpoints are also a bit fussy about which scope the fmtp is declared in
within the SDP.
Adding the fmtp to the SDP is "under construction" in Asterisk. We are
waiting for 1.4 to get underway before we add support for decoding and
encoding fmtp for video and audio. Most of the work is now done, but it
was deemed too different to be put into 1.4. To get things going you
could try and add the fmtp profile, level and maxbr (for instance) to
the SDP in add_codec_to_sdp() with something like...
I think trunk has all those variables available but I've had to change
various things in my build :-( I had this line of code in my builds for
some time before I got around to doing it more fully.
The upcoming support that's needed within asterisk is to make all
these variables configurable (from the conf files) and negotiated across
the various call legs of Asterisk.
I have all of this working, I think, so we should be able to get it
into trunk once 1.4 is out of the way.
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com [mailto:asterisk-video-
bounces@lists.digium.com] On Behalf Of Matteo Brancaleoni
Sent: 19 July 2006 14:14
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] Asterisk H264
I think that I've spotted where the issue is with asterisk
in middle of 2 h264 streams.
H264 requires in SDP the format of the codec. When you make
a call with H264 there's a a:ftmp that specifies at least
the profile-level-id of the h264 stream.
Asterisk should copy that information to the other peer,
in addition to the rtpmap.
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