Posted: Wed Apr 01, 2009 5:11 pm Post subject: [Asterisk-video] app_rtsp: No media found!
Hello,
I'm trying to use Sergio's app_rtsp (latest version) to connect to a
QTSS stream. I'm running Asterisk 1.6.0.
The stream is video only in H.263+ (H.263-1998) encoding. My sip.conf
has "videosupport=yes" and "allow=h263p" among other codecs. We can
successfully have video calls using the X-Lite client using H.263+ for
video. The extension that calls app_rtsp is set up correctly. The
stream should be fine; both Quicktime and VLC play the stream just fine.
When I call up the extension to test app_rtsp I always get "No media
found". It doesn't make it past the DESCRIBE request.
(Also, at the start of the call, there's about 5 to 10 seconds of
repeatedly calling GetUdpPorts before it actually proceeds to the
DESCRIBE. It could be related to these warnings when I compile:
app_rtsp.c: In function 'GetUdpPorts':
app_rtsp.c:284: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
app_rtsp.c:287: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
app_rtsp.c:306: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
...but I don't think this is the source of my problem.)
I've been trying to track down where the problem is... I think it might
be on line 1247:
That if statement evaluates to false for me. The video format gets set
properly to 1048576 (which is the bit for H.263-1998), but the value of
chan->nativeformats is 4, whatever that is, so it's false and those
variables never get set. Later on it decides "no media found". Do I
need to do something to change the value of chan->nativeformats?
I'm attaching my debug log, I added a couple of debug statements of my
own (start with !!!).
Thanks for any help, let me know if I can should any other info,
Mat
<snip ... there's usually about 5-10 seconds of repeated calls to GetUdpPorts here>
Posted: Thu Apr 02, 2009 8:27 am Post subject: [Asterisk-video] app_rtsp: No media found!
Hi Matthew,
If nativeformats is 4, then there is no videosupport for the channel.
Are you calling with xlite?
Try opening the video tab and check that in the SDP there is video
offer. Also try putting the
videosupport=yes on the extension and not only on the general part.
About the GetUdpPorts, RTSP expect two consecutive udp ports, even for
RTP and odd for RTCP,
it should be quite inmediate. Is your machine under heavy load?
It is extrange to have so much difference between two sockets creating
them almos consecutively:
6331,54114,55376,58352,60667,33667,58224,58225
Best regards
Sergio
Matthew Allen escribió:
Quote:
Hello,
I'm trying to use Sergio's app_rtsp (latest version) to connect to a
QTSS stream. I'm running Asterisk 1.6.0.
The stream is video only in H.263+ (H.263-1998) encoding. My sip.conf
has "videosupport=yes" and "allow=h263p" among other codecs. We can
successfully have video calls using the X-Lite client using H.263+ for
video. The extension that calls app_rtsp is set up correctly. The
stream should be fine; both Quicktime and VLC play the stream just fine.
When I call up the extension to test app_rtsp I always get "No media
found". It doesn't make it past the DESCRIBE request.
(Also, at the start of the call, there's about 5 to 10 seconds of
repeatedly calling GetUdpPorts before it actually proceeds to the
DESCRIBE. It could be related to these warnings when I compile:
app_rtsp.c: In function 'GetUdpPorts':
app_rtsp.c:284: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
app_rtsp.c:287: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
app_rtsp.c:306: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
...but I don't think this is the source of my problem.)
I've been trying to track down where the problem is... I think it might
be on line 1247:
That if statement evaluates to false for me. The video format gets set
properly to 1048576 (which is the bit for H.263-1998), but the value of
chan->nativeformats is 4, whatever that is, so it's false and those
variables never get set. Later on it decides "no media found". Do I
need to do something to change the value of chan->nativeformats?
I'm attaching my debug log, I added a couple of debug statements of my
own (start with !!!).
Thanks for any help, let me know if I can should any other info,
Posted: Thu Apr 02, 2009 1:46 pm Post subject: [Asterisk-video] app_rtsp: No media found!
Hi Sergio,
I think that my sip.conf and X-Lite are configured correctly. I tried putting videosupport=yes on the user and not just in [general] but it made no difference. We can have video calls between two X-Lites on this SIP channel and they work.
Interestingly, on subsequent calls my debug log gives the right value for chan->nativeformats:
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1247 rtsp_play: !!! About to check (sdp->video->formats[0]->format & chan->nativeformats) == (1048576 & 1572868) == 1048576
Value 1572868 describes the formats alaw|ulaw|gsm|h263|h263p (as set in my sip.conf).
I'm not sure what caused nativeformats to start giving the correct value, but it was either:
* skipping the check caused some necessary SIP setup to happen
* I tried starting my video in X-Lite while it was doing the GetUdpPorts dance, before app_rtsp, which might have kicked in my video support. (But this was never necessary for normal video calls, or for mp4_save or mp4_play).
However, even though nativeformats is correct now, and it gets past that check, I still see no video in X-Lite.
It seems to set the write format to h263p only (there is no audio in the stream), talks about some rtp packets for a second but then switches to ulaw only for some reason. I never see any video in my client.
Any other clues that might help?
Thanks for any help,
Mat
< snip ... this is near where it used to fail, now it continues ... >
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:749 CreateSDP: -line [a=control:trackID=1]
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1246 rtsp_play: -video [1048576,96,trackID=1]
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1247 rtsp_play: !!! About to check (sdp->video->formats[0]->format & chan->nativeformats) == (1048576 & 1572868) == 1048576
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:510 RtspPlayerSetupVideo: -SETUP VIDEO [trackID=1]
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:562 RtspPlayerPlay: -PLAY [/10-10-24-160-h263.sdp]
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1395 rtsp_play: -Started playback [0]
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3174 ast_rtp_write: Ooh, format changed from unknown to h263p
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 9000, ms is 0 (0), pred/ts/samples 168210/177210/9000
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 9030, ms is 0 (0), pred/ts/samples 276330/285360/9030
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 9030, ms is 0 (0), pred/ts/samples 342390/351420/9030
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 54090, ms is 0 (0), pred/ts/samples 357390/411480/54090
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:1100 ast_rtcp_read: Got RTCP report of 176 bytes
[Apr 2 11:05:00] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 19290, ms is 19 (1710), pred/ts/samples 389160/408450/21000
[Apr 2 11:05:00] DEBUG[16087]: chan_sip.c:5754 sip_rtp_read: Oooh, format changed to 2 gsm
[Apr 2 11:05:00] DEBUG[16087]: channel.c:3376 set_format: Set channel SIP/mat-082168a0 to read format ulaw
[Apr 2 11:05:00] DEBUG[16087]: channel.c:3376 set_format: Set channel SIP/mat-082168a0 to write format ulaw
[Apr 2 11:05:01] DEBUG[16087]: rtp.c:1100 ast_rtcp_read: Got RTCP report of 176 bytes
[Apr 2 11:05:03] DEBUG[16087]: rtp.c:1100 ast_rtcp_read: Got RTCP report of 176 bytes
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com on behalf of Sergio Garcia Murillo
Sent: Thu 4/2/2009 6:16 AM
To: Development discussion of video media support in Asterisk
Subject: Re: [Asterisk-video] app_rtsp: No media found!
Hi Matthew,
If nativeformats is 4, then there is no videosupport for the channel.
Are you calling with xlite?
Try opening the video tab and check that in the SDP there is video
offer. Also try putting the
videosupport=yes on the extension and not only on the general part.
About the GetUdpPorts, RTSP expect two consecutive udp ports, even for
RTP and odd for RTCP,
it should be quite inmediate. Is your machine under heavy load?
It is extrange to have so much difference between two sockets creating
them almos consecutively:
6331,54114,55376,58352,60667,33667,58224,58225
Best regards
Sergio
Matthew Allen escribió:
Quote:
Hello,
I'm trying to use Sergio's app_rtsp (latest version) to connect to a
QTSS stream. I'm running Asterisk 1.6.0.
The stream is video only in H.263+ (H.263-1998) encoding. My sip.conf
has "videosupport=yes" and "allow=h263p" among other codecs. We can
successfully have video calls using the X-Lite client using H.263+ for
video. The extension that calls app_rtsp is set up correctly. The
stream should be fine; both Quicktime and VLC play the stream just fine.
When I call up the extension to test app_rtsp I always get "No media
found". It doesn't make it past the DESCRIBE request.
(Also, at the start of the call, there's about 5 to 10 seconds of
repeatedly calling GetUdpPorts before it actually proceeds to the
DESCRIBE. It could be related to these warnings when I compile:
app_rtsp.c: In function 'GetUdpPorts':
app_rtsp.c:284: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
app_rtsp.c:287: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
app_rtsp.c:306: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
...but I don't think this is the source of my problem.)
I've been trying to track down where the problem is... I think it might
be on line 1247:
That if statement evaluates to false for me. The video format gets set
properly to 1048576 (which is the bit for H.263-1998), but the value of
chan->nativeformats is 4, whatever that is, so it's false and those
variables never get set. Later on it decides "no media found". Do I
need to do something to change the value of chan->nativeformats?
I'm attaching my debug log, I added a couple of debug statements of my
own (start with !!!).
Thanks for any help, let me know if I can should any other info,
Posted: Thu Apr 02, 2009 1:58 pm Post subject: [Asterisk-video] app_rtsp: No media found!
Check h263p is enabled in XLite and get an ethereal trace of the call between Asterisk and Xlite to check if rtp packets are correctly sent from Asterisk.
Best regards
Sergio
Allen, Matthew escribió:
Quote:
Quote:
Hi Sergio,
I think that my sip.conf and X-Lite are configured correctly. I tried putting videosupport=yes on the user and not just in [general] but it made no difference. We can have video calls between two X-Lites on this SIP channel and they work.
Interestingly, on subsequent calls my debug log gives the right value for chan->nativeformats:
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1247 rtsp_play: !!! About to check (sdp->video->formats[0]->format & chan->nativeformats) == (1048576 & 1572868) == 1048576
Value 1572868 describes the formats alaw|ulaw|gsm|h263|h263p (as set in my sip.conf).
I'm not sure what caused nativeformats to start giving the correct value, but it was either:
* skipping the check caused some necessary SIP setup to happen
* I tried starting my video in X-Lite while it was doing the GetUdpPorts dance, before app_rtsp, which might have kicked in my video support. (But this was never necessary for normal video calls, or for mp4_save or mp4_play).
However, even though nativeformats is correct now, and it gets past that check, I still see no video in X-Lite.
It seems to set the write format to h263p only (there is no audio in the stream), talks about some rtp packets for a second but then switches to ulaw only for some reason. I never see any video in my client.
Any other clues that might help?
Thanks for any help,
Mat
< snip ... this is near where it used to fail, now it continues ... >
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:749 CreateSDP: -line [a=control:trackID=1]
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1246 rtsp_play: -video [1048576,96,trackID=1]
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1247 rtsp_play: !!! About to check (sdp->video->formats[0]->format & chan->nativeformats) == (1048576 & 1572868) == 1048576
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:510 RtspPlayerSetupVideo: -SETUP VIDEO [trackID=1]
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:562 RtspPlayerPlay: -PLAY [/10-10-24-160-h263.sdp]
[Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1395 rtsp_play: -Started playback [0]
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3174 ast_rtp_write: Ooh, format changed from unknown to h263p
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 9000, ms is 0 (0), pred/ts/samples 168210/177210/9000
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 9030, ms is 0 (0), pred/ts/samples 276330/285360/9030
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 9030, ms is 0 (0), pred/ts/samples 342390/351420/9030
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 54090, ms is 0 (0), pred/ts/samples 357390/411480/54090
[Apr 2 11:04:59] DEBUG[16087]: rtp.c:1100 ast_rtcp_read: Got RTCP report of 176 bytes
[Apr 2 11:05:00] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 19290, ms is 19 (1710), pred/ts/samples 389160/408450/21000
[Apr 2 11:05:00] DEBUG[16087]: chan_sip.c:5754 sip_rtp_read: Oooh, format changed to 2 gsm
[Apr 2 11:05:00] DEBUG[16087]: channel.c:3376 set_format: Set channel SIP/mat-082168a0 to read format ulaw
[Apr 2 11:05:00] DEBUG[16087]: channel.c:3376 set_format: Set channel SIP/mat-082168a0 to write format ulaw
[Apr 2 11:05:01] DEBUG[16087]: rtp.c:1100 ast_rtcp_read: Got RTCP report of 176 bytes
[Apr 2 11:05:03] DEBUG[16087]: rtp.c:1100 ast_rtcp_read: Got RTCP report of 176 bytes
If nativeformats is 4, then there is no videosupport for the channel.
Are you calling with xlite?
Try opening the video tab and check that in the SDP there is video
offer. Also try putting the
videosupport=yes on the extension and not only on the general part.
About the GetUdpPorts, RTSP expect two consecutive udp ports, even for
RTP and odd for RTCP,
it should be quite inmediate. Is your machine under heavy load?
It is extrange to have so much difference between two sockets creating
them almos consecutively:
6331,54114,55376,58352,60667,33667,58224,58225
Best regards
Sergio
Matthew Allen escribió:
Quote:
Hello,
I'm trying to use Sergio's app_rtsp (latest version) to connect to a
QTSS stream. I'm running Asterisk 1.6.0.
The stream is video only in H.263+ (H.263-1998) encoding. My sip.conf
has "videosupport=yes" and "allow=h263p" among other codecs. We can
successfully have video calls using the X-Lite client using H.263+ for
video. The extension that calls app_rtsp is set up correctly. The
stream should be fine; both Quicktime and VLC play the stream just fine.
When I call up the extension to test app_rtsp I always get "No media
found". It doesn't make it past the DESCRIBE request.
(Also, at the start of the call, there's about 5 to 10 seconds of
repeatedly calling GetUdpPorts before it actually proceeds to the
DESCRIBE. It could be related to these warnings when I compile:
app_rtsp.c: In function 'GetUdpPorts':
app_rtsp.c:284: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
app_rtsp.c:287: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
app_rtsp.c:306: warning: pointer targets in passing argument 3 of
'getsockname' differ in signedness
...but I don't think this is the source of my problem.)
I've been trying to track down where the problem is... I think it might
be on line 1247:
That if statement evaluates to false for me. The video format gets set
properly to 1048576 (which is the bit for H.263-1998), but the value of
chan->nativeformats is 4, whatever that is, so it's false and those
variables never get set. Later on it decides "no media found". Do I
need to do something to change the value of chan->nativeformats?
I'm attaching my debug log, I added a couple of debug statements of my
own (start with !!!).
Thanks for any help, let me know if I can should any other info,
Posted: Thu Apr 02, 2009 4:42 pm Post subject: [Asterisk-video] app_rtsp: No media found!
Good idea Sergio.
In ethereal it looks like for every packet Asterisk gets from QTSS, it
sends one out to X-Lite, in proper H.263+ format. X-Lite isn't
recognizing that it's receiving video. I tried disabling my firewall
completely, no difference.
Something might be wonky with the feature negotiation, since I can have
H.263+ video calls with other X-Lite clients through this Asterisk
instance just fine. So this should be fixable on the server side...
X-Lite was receiving h263p video from other clients and from mp4_play
with no problem.
I'll investigate more and see if I come up with anything.
Thanks for your help,
Mat
X-Lite's debug log doesn't say much about video except that it "can't
locate stream".
Sergio Garcia Murillo wrote:
Quote:
Check h263p is enabled in XLite and get an ethereal trace of the call
between Asterisk and Xlite to check if rtp packets are correctly sent
from Asterisk.
Best regards
Sergio
Allen, Matthew escribió:
> Hi Sergio,
>
> I think that my sip.conf and X-Lite are configured correctly. I tried putting videosupport=yes on the user and not just in [general] but it made no difference. We can have video calls between two X-Lites on this SIP channel and they work.
>
> This morning I tried taking out the check completely, and it seems to get past that point fine now. (Someone else apparently tried this too with the same results: http://www.asteriskguru.com/archives/asterisk-video-amr-audio-continued-vt124127.html )
>
> Interestingly, on subsequent calls my debug log gives the right value for chan->nativeformats:
>
> [Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1247 rtsp_play: !!! About to check (sdp->video->formats[0]->format & chan->nativeformats) == (1048576 & 1572868) == 1048576
>
> Value 1572868 describes the formats alaw|ulaw|gsm|h263|h263p (as set in my sip.conf).
>
> I'm not sure what caused nativeformats to start giving the correct value, but it was either:
> * skipping the check caused some necessary SIP setup to happen
> * I tried starting my video in X-Lite while it was doing the GetUdpPorts dance, before app_rtsp, which might have kicked in my video support. (But this was never necessary for normal video calls, or for mp4_save or mp4_play).
>
> However, even though nativeformats is correct now, and it gets past that check, I still see no video in X-Lite.
>
> It seems to set the write format to h263p only (there is no audio in the stream), talks about some rtp packets for a second but then switches to ulaw only for some reason. I never see any video in my client.
>
> Any other clues that might help?
>
> Thanks for any help,
>
> Mat
>
> < snip ... this is near where it used to fail, now it continues ... >
> [Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:749 CreateSDP: -line [a=control:trackID=1]
> [Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1246 rtsp_play: -video [1048576,96,trackID=1]
> [Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1247 rtsp_play: !!! About to check (sdp->video->formats[0]->format & chan->nativeformats) == (1048576 & 1572868) == 1048576
> [Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:510 RtspPlayerSetupVideo: -SETUP VIDEO [trackID=1]
> [Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:562 RtspPlayerPlay: -PLAY [/10-10-24-160-h263.sdp]
> [Apr 2 11:04:59] DEBUG[16087]: app_rtsp.c:1395 rtsp_play: -Started playback [0]
> [Apr 2 11:04:59] DEBUG[16087]: rtp.c:3174 ast_rtp_write: Ooh, format changed from unknown to h263p
> [Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 9000, ms is 0 (0), pred/ts/samples 168210/177210/9000
> [Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 9030, ms is 0 (0), pred/ts/samples 276330/285360/9030
> [Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 9030, ms is 0 (0), pred/ts/samples 342390/351420/9030
> [Apr 2 11:04:59] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 54090, ms is 0 (0), pred/ts/samples 357390/411480/54090
> [Apr 2 11:04:59] DEBUG[16087]: rtp.c:1100 ast_rtcp_read: Got RTCP report of 176 bytes
> [Apr 2 11:05:00] DEBUG[16087]: rtp.c:3038 ast_rtp_raw_write: Difference is 19290, ms is 19 (1710), pred/ts/samples 389160/408450/21000
> [Apr 2 11:05:00] DEBUG[16087]: chan_sip.c:5754 sip_rtp_read: Oooh, format changed to 2 gsm
> [Apr 2 11:05:00] DEBUG[16087]: channel.c:3376 set_format: Set channel SIP/mat-082168a0 to read format ulaw
> [Apr 2 11:05:00] DEBUG[16087]: channel.c:3376 set_format: Set channel SIP/mat-082168a0 to write format ulaw
> [Apr 2 11:05:01] DEBUG[16087]: rtp.c:1100 ast_rtcp_read: Got RTCP report of 176 bytes
> [Apr 2 11:05:03] DEBUG[16087]: rtp.c:1100 ast_rtcp_read: Got RTCP report of 176 bytes
>
>
>
>
> -----Original Message-----
> From: asterisk-video-bounces@lists.digium.com on behalf of Sergio Garcia Murillo
> Sent: Thu 4/2/2009 6:16 AM
> To: Development discussion of video media support in Asterisk
> Subject: Re: [Asterisk-video] app_rtsp: No media found!
>
> Hi Matthew,
>
> If nativeformats is 4, then there is no videosupport for the channel.
> Are you calling with xlite?
> Try opening the video tab and check that in the SDP there is video
> offer. Also try putting the
> videosupport=yes on the extension and not only on the general part.
>
> About the GetUdpPorts, RTSP expect two consecutive udp ports, even for
> RTP and odd for RTCP,
> it should be quite inmediate. Is your machine under heavy load?
> It is extrange to have so much difference between two sockets creating
> them almos consecutively:
>
> 6331,54114,55376,58352,60667,33667,58224,58225
>
> Best regards
> Sergio
>
> Matthew Allen escribió:
>
>> Hello,
>>
>> I'm trying to use Sergio's app_rtsp (latest version) to connect to a
>> QTSS stream. I'm running Asterisk 1.6.0.
>>
>> The stream is video only in H.263+ (H.263-1998) encoding. My sip.conf
>> has "videosupport=yes" and "allow=h263p" among other codecs. We can
>> successfully have video calls using the X-Lite client using H.263+ for
>> video. The extension that calls app_rtsp is set up correctly. The
>> stream should be fine; both Quicktime and VLC play the stream just fine.
>>
>> When I call up the extension to test app_rtsp I always get "No media
>> found". It doesn't make it past the DESCRIBE request.
>>
>> (Also, at the start of the call, there's about 5 to 10 seconds of
>> repeatedly calling GetUdpPorts before it actually proceeds to the
>> DESCRIBE. It could be related to these warnings when I compile:
>>
>> app_rtsp.c: In function 'GetUdpPorts':
>> app_rtsp.c:284: warning: pointer targets in passing argument 3 of
>> 'getsockname' differ in signedness
>> app_rtsp.c:287: warning: pointer targets in passing argument 3 of
>> 'getsockname' differ in signedness
>> app_rtsp.c:306: warning: pointer targets in passing argument 3 of
>> 'getsockname' differ in signedness
>>
>> ...but I don't think this is the source of my problem.)
>>
>> I've been trying to track down where the problem is... I think it might
>> be on line 1247:
>>
>> 1247: if (sdp->video->formats[i]->format & chan->nativeformats)
>> 1248: {
>> < ... sets videoType, videoFormat, videoControl ...>
>> 1257: }
>>
>> That if statement evaluates to false for me. The video format gets set
>> properly to 1048576 (which is the bit for H.263-1998), but the value of
>> chan->nativeformats is 4, whatever that is, so it's false and those
>> variables never get set. Later on it decides "no media found". Do I
>> need to do something to change the value of chan->nativeformats?
>>
>> I'm attaching my debug log, I added a couple of debug statements of my
>> own (start with !!!).
>>
>> Thanks for any help, let me know if I can should any other info,
>>
>> Mat
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-video
>>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
You can post new topics in this forum You can 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
FAQ
Search
Memberlist
Usergroups
Register
Profile
Log in to check your private messages
Log in
