[Asterisk-video] Asterisk 11.6 h.264 passthrought Polycom VVX 500

David Gagnon gagnon.david at gmail.com
Mon Nov 11 12:28:06 CST 2013


Hi,

H.264 won't appear in core show translation because Asterisk is not able to
transcode the video. But with core show codecs :

voip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
        It does not indicate anything about your configuration.
      ID  TYPE     NAME DESCRIPTION
-----------------------------------------------------------------------------------
  100001 audio     g723 (G.723.1)
  100002 audio      gsm (GSM)
  100003 audio     ulaw (G.711 u-law)
  100004 audio     alaw (G.711 A-law)
  100011 audio     g726 (G.726 RFC3551)
  100006 audio    adpcm (ADPCM)
  100019 audio     slin (16 bit Signed Linear PCM)
  100007 audio    lpc10 (LPC10)
  100008 audio     g729 (G.729A)
  100009 audio    speex (SpeeX)
  100016 audio  speex16 (SpeeX 16khz)
  100010 audio     ilbc (iLBC)
  100005 audio g726aal2 (G.726 AAL2)
  100012 audio     g722 (G722)
  100021 audio   slin16 (16 bit Signed Linear PCM (16kHz))
  300001 image     jpeg (JPEG image)
  300002 image      png (PNG image)
  200001 video     h261 (H.261 Video)
  200002 video     h263 (H.263 Video)
  200003 video    h263p (H.263+ Video)
  200004 video     h264 (H.264 Video)
  200005 video    mpeg4 (MPEG4 Video)
  400001  text      red (T.140 Realtime Text with redundancy)
  400002  text     t140 (Passthrough T.140 Realtime Text)
  100013 audio   siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
  100014 audio  siren14 (ITU G.722.1 Annex C, (Siren14, licensed from
Polycom))
  100017 audio  testlaw (G.711 test-law)
  100015 audio     g719 (ITU G.719)
  100028 audio  speex32 (SpeeX 32khz)
  100020 audio   slin12 (16 bit Signed Linear PCM (12kHz))
  100022 audio   slin24 (16 bit Signed Linear PCM (24kHz))
  100023 audio   slin32 (16 bit Signed Linear PCM (32kHz))
  100024 audio   slin44 (16 bit Signed Linear PCM (44kHz))
  100025 audio   slin48 (16 bit Signed Linear PCM (48kHz))
  100026 audio   slin96 (16 bit Signed Linear PCM (96kHz))
  100027 audio  slin192 (16 bit Signed Linear PCM (192kHz))

My problem is that the callee don't have the video but the caller do. So
Asterisk is able to passthrought the h.264 codec in one direction.

David


On Mon, Nov 11, 2013 at 10:17 AM, || dave cantera Mobile <
davidjcantera at gmail.com> wrote:

>  david,
> then you might want to check the module or codec list. perhaps the h.264
> codec is not being loaded, remember new versions of * changed the cmd
> structure, it might be core show codecs now.
> daveC
>
> http://www.voip-info.org/wiki/view/Asterisk+codecs
>
> Use this commands in the Asterisk CLI for a detailed listing of the actual
> capabilities:
>
> show codecs **
> show translation
> show translation recalc 10
>
>
>
>
>     - show codecs Screen output
>
> The 'show codecs' command is deprecated and will be removed in a future
> release. Please use 'core show codecs' instead.
>
> 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)
>
>
>
>
>
> On 11/11/2013 7:12 AM, David Gagnon wrote:
>
> Hi,
>
>  I'm on a local network for the moment, no nat. I can get h.263 to work
> no problem but not h.264. There is no firewall involved. It'S seem to be
> related to h.264 for some reason.
>
>  David
>
>
> On Sun, Nov 10, 2013 at 12:09 AM, || dave cantera Mobile <
> davidjcantera at gmail.com> wrote:
>
>>  david,
>> check the firewall settings for port blocking tcp/udp open both ways for
>> 10000-20000, sounds like router is not letting traffic through
>>
>>
>> On 11/9/2013 7:56 PM, David Gagnon wrote:
>>
>> Hi,
>>
>>  I'm running Asterisk 11.6 and I'm unable to get h.264 to work with two
>> Polycom VVX  500 running UC 5.0.1. If I enable h263 only in sip.conf it's
>> working no problem but if I enable h.264 only it's not worning. I was able
>> to get h.264 to work if I enable directmedia=yes. In my lab this is working
>> but in production, both phone will be behind a different NAT. Have anyone
>> been able to use h.264 passthrought with Asterisk 11 and directmedia=no ?
>>
>>  The problem is that that caller don't get the video from the callee but
>> the callee get the video of the caller. It's sort of one-way video.
>>
>>  Thx !
>>
>>
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-video/attachments/20131111/d7f2e3bc/attachment-0001.html>


More information about the asterisk-video mailing list