thr3ads.net - asterisk users - [asterisk-users] Asterisk not using common codec between (SIP) endpoints [Oct 2019]

If this information is useful, please help other people find it:
Share via:

Andreas Wehrmann

2019-Oct-03 11:13 UTC

[asterisk-users] Asterisk not using common codec between (SIP) endpoints

Hello people,

I've ran into two problem that I can't seem to be able to solve on my
own.
Here's my scenario (running Asterisk 13.28.1):

In short: - Asterisk behaves unexpectedly (at least to me) when 
negotiating between endpoints
             that have a different but intersecting set of codecs 
(preventing direct media flow).

           - Also, when an endpoint sends RTP with an unexpected (but 
internally hardcoded) payload type,
             Asterisk seems to get "confused" with what codec is 
supposed to be active (which seems to break transcoding).

I've got a couple of phones (supported codecs are: g722,alaw,ulaw (in 
that order))
and a "special" media gateway which supports alaw, ulaw only.
What I mean by "special" is, that it sends status information via RTP 
payload type 102;
this is hardcoded and I cannot change it; also that thing sends it 
regardless
of what was negotiated via SDP (yes, it's stupid and annoying to deal with).

What I actually wanted to test was to see whether Asterisk would 
correctly negotiate codecs
when making calls between the different kinds of participants (phones 
and gateways).

Here's what I tested:

- Calls between phones (OK)
-- phone01 calls phone02 (phone01 offer: g722, alaw, ulaw)
-- phone02 accepts (Asterisk/phone02 offer/answer: g722, alaw, ulaw)
-- Asterisk sends reinvite to both to establish direct media flow 
(codec: g722)
-- OKAY - this is exactly what i would expect

- Phone calls mediagateway; mediagateway sends special RTP (NOK: 
Asterisk attempts to transcode and RTP engine seems to get confused)
-- phone01 calls mediagateway (phone01 offer: g722, alaw, ulaw)
-- mediagateway accepts call (Asterisk offer: alaw, ulaw | mediagateway 
answer: alaw)
-- Asterisk accepts call from phone01 (Asterisk answer: g722, alaw, ulaw)
-- Asterisk attempts to transcodes between g722 and alaw

--> My expectation here would be for Asterisk to reduce the set of 
codecs in the answer to phone01 to alaw only,
     since this would enable direct media flow between the endpoints.
     But even if it accepts the call like I described above,
     I would expect Asterisk to reinvite the calls later
     to use alaw only and get out of the way to enable direct media flow.

-- After mediagateway accepts the call, it sends a few status messages 
via RTP PT 102;
    this seems to confuse the RTP engine in Asterisk (see the console 
output at the end of this mail)
    which also seems to break transcoding, because audio doesn't work in 
either direction.


My questions are:

- Is there a way to make Asterisk always try to get out of the media path?
   I thought direct_media=yes and an overlapping set of codecs would 
suffice.

- Even if direct_media is disabled: Is there a way to make Asterisk 
always use a common codec between SIP endpoints,
   so it doesn't need to transcode?

- When Asterisk receives unexpected RTP packets or RTP packets of an 
unexpected type in a session,
   shouldn't it just ignore them?
   This is what I observed in older Asterisk versions (very old, like 1.4)
   where it would simply print a warning about an unknown/unhandled 
payload type.

I'll attach my pjsip.conf, extensions.ael and a tcpdump that shows my 
phone (10.137.8.20)
calling the mediagateway (10.254.0.221) via Asterisk (10.137.8.19).
The rest of the configs are the default configs created with "make 
basic-pbx".

I do hope, I'm not missing something obvious here...

A few additions: I ran the tests with Asterisk 16.5.1 also, the results are:

- Calls between phones work as expected, G722 is used and Asterisks 
reinvites
   to establish direct media flow between the phones.

- When a phone calls a mediagateway, the results are as above with the 
exception of having audio in both directions.
   The status messages sent via RTP PT 102 do not seem to confuse 
Asterisk 16.5.1.

- I also ran this test: I have two endpoints (direct_media=yes) each 
with allowed codecs
   set to ulaw, alaw on one endpoint and alaw, ulaw (reverse order) set 
on the other endpoint.
   When endpoint A calls B and offers codecs (in that order) ulaw, alaw 
and endpoint B accepts alaw only,
   Asterisk uses ulaw for the call from A and alaw in the call to B.

Like I said above: I really hope I'm not missing something obvious.
I need Asterisk to use the same codec in both calls (if there is an 
overlapping set)
and try to get out of the way by establishing a direct media path if 
somehow possible.


Best Regards,
Andreas



NOTE: The following is the console output from Asterisk 13 when the 
mediagateway answers and sends RTP PT 102 at the beginning of the call.

[Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 
ast_translator_build_path: No translator path: (starting codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 
ast_translator_build_path: No translator path: (starting codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 
ast_translator_build_path: No translator path: (starting codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:56] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:56] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:56] WARNING[23961][C-00000000]: chan_pjsip.c:856 
chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw 
frame when native formats are (siren7) 
(rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) 
wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000))
[Oct  2 07:24:56] WARNING[23961][C-00000000]: channel.c:5751 set_format: 
Unable to find a codec translation path: (siren7) -> (alaw)
[Oct  2 07:24:56] WARNING[23961][C-00000000]: channel.c:5751 set_format: 
Unable to find a codec translation path: (alaw) -> (siren7)

-------------- next part --------------


context internal {
    96 => {
        Dial(PJSIP/boslwzldi21);
    }
    100 => {
        Dial(PJSIP/andytel);
    }
    102 => {
        Dial(PJSIP/boslwztel02);
    }
}
-------------- next part --------------
A non-text attachment was scrubbed...
Name: phone_calls_mediagateway_sip_rtp_asterisk13.pcap
Type: application/vnd.tcpdump.pcap
Size: 8179 bytes
Desc: not available
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20191003/c242c99c/attachment.pcap>
-------------- next part --------------
;================================ TRANSPORTS =[transport-udp]
type = transport
protocol = udp
bind = 10.137.8.19

;================================ ENDPOINTS =
[endpoint-basic]
type=endpoint
context = internal
disallow = all
allow = g722,alaw,ulaw
direct_media = yes

[endpoint-radio]
type=endpoint
context = internal
disallow = all
allow = alaw,ulaw
direct_media = yes

[auth-userpass](!)
type = auth
auth_type = userpass
password = SnomSecret
realm = A

[aor-single-reg](!)
type = aor
max_contacts = 1
remove_existing = yes
qualify_frequency = 60

;============== RADIO GATEWAYS
[boslwzldi21](endpoint-radio)
aors=boslwzldi21

[boslwzldi21](aor-single-reg)
contact=sip:10.254.0.221

[boslwzldi21]
type = identify
endpoint = boslwzldi21
match = 10.254.0.221

;============= SNOM
[andytel](endpoint-basic)
;auth=andytel
aors=andytel

[andytel](aor-single-reg)

[andytel](auth-userpass)
username=andytel

[boslwztel02](endpoint-basic)
;auth=auth-userpass
aors=boslwztel02

[boslwztel02](aor-single-reg)

[boslwztel02](auth-userpass)
username=boslwztel02

Administrator TOOTAI

2019-Oct-03 13:08 UTC

head link

[asterisk-users] Asterisk not using common codec between (SIP) endpoints

Hi

Le 03/10/2019 à 13:13, Andreas Wehrmann a écrit :

[...]> 
> - Even if direct_media is disabled: Is there a way to make Asterisk 
> always use a common codec between SIP endpoints,
>    so it doesn't need to transcode?
Before calling the gatreway add

same = n,set(SIP_CODEC=alaw)

[...]

-- 
Daniel

Andreas Wehrmann

2019-Oct-03 14:10 UTC

head link

[asterisk-users] Asterisk not using common codec between (SIP) endpoints

On 03.10.19 15:08, Administrator TOOTAI wrote:
> Before calling the gatreway add
>
> same = n,set(SIP_CODEC=alaw)
>
> [...]
>
Hey there,

that doesn't work as it seems to be implemented for chan_sip only;
I'm using chan_pjsip; sorry if I didn't explain myself properly.

Anyway, in my case that would not really be an acceptable solution anyway,
because I need the called party to be able to pick from the range of 
codecs presented to it
because the codec chosen by the destination might change (my example is 
a simplified version).

I don't think putting the burden of worrying about audio codecs on the 
dialplan writer is a good idea,
since this should be dealt with automatically with respect to what is 
configured and negotiated.
This is also because in the systems I have to work with, the 'engineers'
usual provide the configuration (endpoints, NAT config and the like)
while the technicians implement the dialplan (or the business logic so 
to speak) according to customer needs.
They (the technicians) usually don't know (much) about codecs or how the 
channels techs work exactly...

Thanks,
Andy

Reasonably Related Threads

Search for more possibly parallel threads

asterisk users - Oct 2019 - Asterisk not using common codec between (SIP) endpoints

[asterisk-users] Asterisk not using common codec between (SIP) endpoints

[asterisk-users] Asterisk not using common codec between (SIP) endpoints

[asterisk-users] Asterisk not using common codec between (SIP) endpoints

Reasonably Related Threads