asterisk users - Dec 2013 - Why doesn't Asterisk try to prevent transcoding

Let's say I have two devices configured and the follow call scenarios occur.

[100]
disallow=all
allow=g722&ulaw

Polycom phone with g722,ulaw,alaw,g729

[101]
disallow=all
allow=ulaw

Polycom phone with g722,ulaw,alaw,g729

101 dials 100 -> ulaw to ulaw is chosen
100 dials 101 -> g722 to ulaw is chosen

Ideally when 100 dials 101 ulaw would be chosen since it is the common
format. Looking into this deeper

Device 100 sends INVITE to Asterisk offering g722,ulaw,alaw,g729
Asterisk sends INVITE to device 101 offering ulaw
Device 101 sends 200 OK to Asterisk offering ulaw
Asterisk sends 200 OK to device 100 offering g722,ulaw

I can prevent transcoding by adding SIP_CODEC_INBOUND=ulaw to the dialplan
for extension 101. This causes Asterisk to send 200 OK to device 100
offering ulaw. Am I missing why Asterisk wouldn't just offer the highest
priority codec they have in common to prevent transcoding?

Ryan
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20131214/28209c12/attachment.html>

On Sat, Dec 14, 2013 at 10:31 PM, Ryan Wagoner <rswagoner at gmail.com>
wrote:
> Let's say I have two devices configured and the follow call scenarios
> occur.
>
> [100]
> disallow=all
> allow=g722&ulaw
>
> Polycom phone with g722,ulaw,alaw,g729
>
> [101]
> disallow=all
> allow=ulaw
>
> Polycom phone with g722,ulaw,alaw,g729
>
> 101 dials 100 -> ulaw to ulaw is chosen
> 100 dials 101 -> g722 to ulaw is chosen
>
> Ideally when 100 dials 101 ulaw would be chosen since it is the common
> format. Looking into this deeper
>
> Device 100 sends INVITE to Asterisk offering g722,ulaw,alaw,g729
> Asterisk sends INVITE to device 101 offering ulaw
> Device 101 sends 200 OK to Asterisk offering ulaw
> Asterisk sends 200 OK to device 100 offering g722,ulaw
>
> I can prevent transcoding by adding SIP_CODEC_INBOUND=ulaw to the dialplan
> for extension 101. This causes Asterisk to send 200 OK to device 100
> offering ulaw. Am I missing why Asterisk wouldn't just offer the
highest
> priority codec they have in common to prevent transcoding?
>
> Ryan
>
I should have mentioned I'm using Asterisk 11.2-cert2. The core debug from
the above shows

[2013-12-14 22:51:59] DEBUG[25200][C-0000004d]: chan_sip.c:7911 sip_new:
*** Our native formats are (g722)
[2013-12-14 22:51:59] DEBUG[25200][C-0000004d]: chan_sip.c:7912 sip_new:
*** Joint capabilities are (ulaw|g722)
[2013-12-14 22:51:59] DEBUG[25200][C-0000004d]: chan_sip.c:7913 sip_new:
*** Our capabilities are (ulaw|g722)
[2013-12-14 22:51:59] DEBUG[25200][C-0000004d]: chan_sip.c:7914 sip_new:
*** AST_CODEC_CHOOSE formats are g722

[2013-12-14 22:51:59] DEBUG[27830][C-0000004d]: chan_sip.c:7911 sip_new:
*** Our native formats are (ulaw)
[2013-12-14 22:51:59] DEBUG[27830][C-0000004d]: chan_sip.c:7912 sip_new:
*** Joint capabilities are (nothing)
[2013-12-14 22:51:59] DEBUG[27830][C-0000004d]: chan_sip.c:7913 sip_new:
*** Our capabilities are (ulaw)
[2013-12-14 22:51:59] DEBUG[27830][C-0000004d]: chan_sip.c:7914 sip_new:
*** AST_CODEC_CHOOSE formats are ulaw
[2013-12-14 22:51:59] DEBUG[27830][C-0000004d]: chan_sip.c:7916 sip_new:
*** Our preferred formats from the incoming channel are (g722)

I'm looking at the code now. I am hoping to write a patch, if I can wrap my
head around the code, to determine join capabilities between the joint
capabilities of each channel. If this exists then set both channels this
codec.

Ryan
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20131214/a399302c/attachment.html>

I think the order or elements is relevant:

[100]
disallow=all
allow=ulaw
allow=g722
or
[100]
allow=!all,ulaw,g722

should work.

jg

I have had the issue for years. The problem is that Asterisk
developers are removed from the business. We desperately need simple
way to eliminate transcoding when unnecessary. Transcoding brings a
server to its knees. It is a very simple new setting in sip.conf
prioritize_matching_codecs=yes
I vote for this new feature. However, I don't have the expertise to
write  a patch. I would say that only Digium developers could attempt
to do this without disrupting the code too much. I also tried to
migrate to PJSIP, but had to go back when I realized there was no
channel variable contaning the inbound IP address. In general, any
channel hast to provide the information to the dialplan, somehow,
otherwise we cannot do business. I hope the PJSIP integration matures
soon.