Posted: Thu Jan 17, 2008 11:35 am Post subject: [Asterisk-video] announcing bandwidth in SDP
Great, but does EyeBeam comfom the strictly the traffic to the desired bandwith??
I'll be very interested in knowing the results, if you got a video recorded that way send it to me, I'll try to take a look at the video stream to see how "good" it's for directly using it with the videocall.
Best regards
Sergio
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
To: asterisk-video@lists.digium.com
Sent: Thu, 17 Jan 2008 12:17:35 +0100
Subject: Re: [Asterisk-video] announcing bandwidth in SDP
Klaus Darilion schrieb:
Quote:
Hi!
Is there an Asterisk version (trunk, videocaps branch ...) which allows
to signal the maximum video bandwidth (e.g. b= or a=fmtp:...MaxBR=)
Answering myself: there is maxcallbitrate in sip.conf. It adds b=CS: to
global section of SDP.
Looks like it is ignored by eyebeam, which uses MaxBr setting in a=ftmp
line which was defined in a rather old SDP draft.
regards
klaus
_______________________________________________
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Posted: Thu Jan 17, 2008 12:09 pm Post subject: [Asterisk-video] announcing bandwidth in SDP
First test:
I added the same string as eyebeam adds in the "Cable, DSL" setting to
the SDP: a=fmtp:103 QCIF=1 I=1 J=1 K=1 MaxBR=4520
Result: The 3G handset did not showed video at all.
Then I added only the MaxBR setting:
a=fmtp:103 QCIF=1 MaxBR=800
This works fine an Eyebeam video is diplayed on the 3G handset.
Values >800 improve video quality, but video delay is steadily
increasing (shouldn't there be some packets droped to avoid to big delays?)
Values <800 improves delay (video is getting real fast) but decreases
quality.
regards
klaus
Sergio Garcia Murillo schrieb:
Quote:
Great, but does EyeBeam comfom the strictly the traffic to the desired bandwith??
I'll be very interested in knowing the results, if you got a video recorded that way send it to me, I'll try to take a look at the video stream to see how "good" it's for directly using it with the videocall.
Best regards
Sergio
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
To: asterisk-video@lists.digium.com
Sent: Thu, 17 Jan 2008 12:17:35 +0100
Subject: Re: [Asterisk-video] announcing bandwidth in SDP
Klaus Darilion schrieb:
> Hi!
>
> Is there an Asterisk version (trunk, videocaps branch ...) which allows
> to signal the maximum video bandwidth (e.g. b= or a=fmtp:...MaxBR=)
Answering myself: there is maxcallbitrate in sip.conf. It adds b=CS: to
global section of SDP.
Looks like it is ignored by eyebeam, which uses MaxBr setting in a=ftmp
line which was defined in a rather old SDP draft.
regards
klaus
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Posted: Thu Jan 17, 2008 2:35 pm Post subject: [Asterisk-video] announcing bandwidth in SDP
Part 2: Attached is the patch for chan_sip if someone wants to try it.
You can specify the bandwidth with
maxcallbitrate_h263p=value
where value is the bandwidth in kbit/s. Note: this only works in the
global section. value=80 is a good value for H324M calls.
The added SDP line will be:
a=fmtp:103 QCIF=1 MaxBR=value0
(maxbr is bit/100s)
If 0 then no SDP line will be added.
regards
klaus
Sergio Garcia Murillo schrieb:
Quote:
Great, but does EyeBeam comfom the strictly the traffic to the desired bandwith??
I'll be very interested in knowing the results, if you got a video recorded that way send it to me, I'll try to take a look at the video stream to see how "good" it's for directly using it with the videocall.
Best regards
Sergio
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
To: asterisk-video@lists.digium.com
Sent: Thu, 17 Jan 2008 12:17:35 +0100
Subject: Re: [Asterisk-video] announcing bandwidth in SDP
Klaus Darilion schrieb:
> Hi!
>
> Is there an Asterisk version (trunk, videocaps branch ...) which allows
> to signal the maximum video bandwidth (e.g. b= or a=fmtp:...MaxBR=)
Answering myself: there is maxcallbitrate in sip.conf. It adds b=CS: to
global section of SDP.
Looks like it is ignored by eyebeam, which uses MaxBr setting in a=ftmp
line which was defined in a rather old SDP draft.
regards
klaus
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
--- asterisk-1.4.17/configs/sip.conf.sample 2007-11-27 08:34:19.000000000 +0100
+++ asterisk-1.4.17-h324m/configs/sip.conf.sample 2008-01-17 15:07:07.000000000 +0100
@@ -123,6 +123,11 @@
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
; for peers and users as well
+;maxcallbitrate_h263p=80 ; Maximum bitrate for video calls. 0 means 'no maximum'
+ ; thus no header is added to SDP. (default 0 kb/s)
+ ; This is only a global setting. This option uses the obsolete
+ ; MaxBR parameter. Successful tested with eyebeam.
+ ; 80 kbit is fine for SIP<-->H324M calls
;callevents=no ; generate manager events when sip ua
; performs events (e.g. hold)
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
--- asterisk-1.4.17/channels/chan_sip.c 2008-01-02 21:24:09.000000000 +0100
+++ asterisk-1.4.17-h324m/channels/chan_sip.c 2008-01-17 15:18:51.000000000 +0100
@@ -510,6 +510,7 @@
#define DEFAULT_QUALIFY FALSE
#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
+#define DEFAULT_MAX_CALL_BITRATE_H263P (0) /*!< Max bitrate for video for h263p codec */
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
#endif
@@ -528,7 +529,8 @@
static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
* a bridged channel on hold */
-static int default_maxcallbitrate; /*!< Maximum bitrate for call */
+static int default_maxcallbitrate; /*!< Maximum bitrate for call */
+static int default_maxcallbitrate_h263p; /*!< Maximum bitrate for call for h263p codec */
static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
/* Global settings only apply to the channel */
@@ -960,6 +962,7 @@
int jointnoncodeccapability; /*!< Joint Non codec capability */
int redircodecs; /*!< Redirect codecs */
int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
+ int maxcallbitrate_h263p; /*!< Maximum Call Bitrate for Video Calls with h263p codec*/
struct t38properties t38; /*!< T38 settings */
struct sockaddr_in udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */
struct ast_udptl *udptl; /*!< T.38 UDPTL session */
@@ -4463,6 +4466,7 @@
if (p->udptl)
ast_udptl_settos(p->udptl, global_tos_audio);
p->maxcallbitrate = default_maxcallbitrate;
+ p->maxcallbitrate_h263p = default_maxcallbitrate_h263p;
}
if (useglobal_nat && sin) {
@@ -6180,6 +6184,10 @@
} else if (codec == AST_FORMAT_ILBC) {
/* Add information about us using only 20/30 ms packetization */
ast_build_string(a_buf, a_size, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
+ } else if (codec == AST_FORMAT_H263_PLUS) {
+ /* Add eyebeam style maxbandwidth description */
+ if (p->maxcallbitrate_h263p)
+ ast_build_string(a_buf, a_size, "a=fmtp:%d QCIF=1 MaxBR=%d0\r\n", rtp_code,p->maxcallbitrate_h263p);
}
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