[asterisk-users] [External] [External] [External] Asterisk 18.12.0 question

[Dan Cropp]

After further testing, not sure this is chan_sip related.

I can disable chan_sip.so from loading in modules.conf and that does solve the startup/loading for res_pjsip_transport_websocket.
However, there is some issue with the wss transport.  Seeing this in both 16.26.0 (not in 16.25.0) and 18.12.0 (not in 18.11.2).

REGISTER message comes in, is accepted.  However, when it goes to send the OPTIONS, it’s outputting the Unsupported transport.


[05/19 10:11:41.992] VERBOSE[2456] res_pjsip_logger.c: <--- Received SIP request (907 bytes) from WSS:192.168.32.27:56443 --->
REGISTER sip:mybox.mydomain.com SIP/2.0
Via: SIP/2.0/WSS c2537bthsnvo.invalid;branch=z9hG4bK2816987
Max-Forwards: 69
To: <sip:1234 at mybox.mydomain.com>
From: <sip:1234 at mybox.mydomain>;tag=24ipeon952
Call-ID: lshogr91tba8r5f335c1g5
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="1234", realm="asterisk", nonce="1652973101/72159fe10d9432b64a16fec84fc414e7", uri="sip:mybox.mydomain.com", response="f46f710af7db6e2e86ec2fabe38325e8", opaque="06a146a816d699e2", qop=auth, cnonce="meehpb38l93l", nc=00000001
Contact: <sip:e6hj0uh4 at c2537bthsnvo.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:032f947e-da85-4920-b944-86b52760937b>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 3.3.6
Content-Length: 0


[05/19 10:11:41.993] VERBOSE[2456] res_pjsip_logger.c: <--- Transmitting SIP response (482 bytes) to WSS:192.168.32.27:56443 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS c2537bthsnvo.invalid;rport=56443;received=192.168.32.27;branch=z9hG4bK2816987
Call-ID: lshogr91tba8r5f335c1g5
From: <sip:1234 at mybox.mydomain.com>;tag=24ipeon952
To: <sip:1234 at mybox.mydomain.com>;tag=z9hG4bK2816987
CSeq: 2 REGISTER
Date: Thu, 19 May 2022 15:11:41 GMT
Contact: <sip:e6hj0uh4 at 192.168.32.27:56443;transport=ws>;expires=599
Expires: 600
Server: Asterisk PBX 18.12.0
Content-Length:  0


[05/19 10:11:41.994] ERROR[2456] res_pjsip.c: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint 1234


Identical behavior happening with Asterisk 16.26.0, but not on Asterisk 16.25.0
Configuration files are same for between Asterisk versions.

[transport3]
type = transport
bind = 0.0.0.0
protocol = wss
allow_reload = no

[1234]
type = aor
max_contacts = 1
remove_existing = yes
qualify_frequency = 60

[1234]
type = auth
auth_type = userpass
username = 1234
password = mypassword

[1234]
type = endpoint
context = IS
auth = 1234
aors = 1234
dtmf_mode = rfc4733
webrtc = yes
disallow = all
allow = ulaw
transport = transport3
acl = acl5


Might this be because PJSIP 2.12 changes to the
“WebRTC updates with AEC3 & AGC2”



From: Dan Cropp
Sent: Friday, May 13, 2022 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Subject: RE: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 question

Thank you Joshua!!!

Not loading chan_sip module resolved the problem.

Hope you have an awesome weekend.

From: asterisk-users <asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>> On Behalf Of Joshua C. Colp
Sent: Friday, May 13, 2022 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com<mailto:asterisk-users at lists.digium.com>>
Subject: Re: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 question

On Fri, May 13, 2022 at 3:19 PM Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote:
Thanks Joshua.

I didn’t describe that very well.

When I first noticed the res_http_transport_websocket wasn’t loading on that box, I compared the modules folder on both boxes.  My thought was I forgot some module that was required.

I noticed I forgot to include these files, so I added them to the package.  Rolled back the VM and re-installed.  Didn’t make a difference whether they were present or not.
/usr/lib/asterisk/modules/codec_g729a.*
/usr/lib/asterisk/modules/codec_silk.*
/usr/lib/asterisk/modules/codec_siren14.*
/usr/lib/asterisk/modules/codec_siren7.*
/usr/lib/asterisk/modules/format_ogg_opus.so

Comparing the menuselect-tree between the two versions, only changes I see are
func_evalexten
res_aeap
res_speech_aeap
and four test_aeap_... added to the TEST_FRAMEWORK.

Would it make sense for me to modify my bash script to disable those settings, compile, and try installing?  Bash script configures the menuselect options and compiles asterisk.
Seems like that would be a better apples to apples comparison.  Eliminating the new features.

You can. It would also make sense as a test to just not load chan_sip and see what happens.

--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org>
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