Call without m= attribute in SDP

Asterisk Asterisk Support
1

Hello.
I am trying to connect door phone intercom Hikvision DS-KV8102-IP with Asterisk 13.1.0~dfsg-1.1ubuntu4.1.
In configure menu of this intercom is possible set only the ip address of SIP server. From sniffs of comunication bettwen intercom and asterisk I prepared one peer with following settings:

[againdevice]
fullname = Door intercom
username = againdevice
cid_number = 11
insecure=invite
qualify=yes
type=friend
host=0.0.0.0
allowguest=yes
allow=all
directmedia=yes
directrtpsetup=yes
progressinband=no
ignoresdpversion=yes
prematuremedia=yes

But intercom does not send m= attribute in SDP content of INVITE packet and the call is rejected with “SIP/2.0 488 Not acceptable here”.

<--- SIP read from UDP:192.168.35.222:5060 --->
INVITE sip:againdevice@192.168.35.116:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.35.222:5060;rport;branch=z9hG4bK1248230690
From: <sip:againdevice@192.168.35.222>;tag=1833694817
To: <sip:againdevice@192.168.35.116:5060>
Call-ID: 88674963
CSeq: 20 INVITE
Contact: <sip:againdevice@192.168.35.222:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: eXosip/3.6.0
Subject: This is a call for conversation
Content-Length: 139

v=0
o=649788304 0 0 IN IP4 192.168.35.222
s=Talk session
c=IN IP4 192.168.35.222
t=0 0
a=doorFloor:6
a=responseType:0
a=doorType:0
<------------->
--- (12 headers 8 lines) ---
Sending to 192.168.35.222:5060 (no NAT)
Sending to 192.168.35.222:5060 (no NAT)
Using INVITE request as basis request - 88674963
Found peer 'againdevice' for 'againdevice' from 192.168.35.222:5060
  == Using SIP RTP CoS mark 5
[Aug 15 09:18:04] WARNING[17232][C-00000005]: chan_sip.c:9970 process_sdp: Insufficient information for SDP (m= not found)

<--- Reliably Transmitting (NAT) to 192.168.35.222:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.35.222:5060;branch=z9hG4bK1248230690;received=192.168.35.222;rport=5060
From: <sip:againdevice@192.168.35.222>;tag=1833694817
To: <sip:againdevice@192.168.35.116:5060>;tag=as7ca216c6
Call-ID: 88674963
CSeq: 20 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '88674963' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.35.222:5060 --->
ACK sip:againdevice@192.168.35.116:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.35.222:5060;rport;branch=z9hG4bK1248230690
Route: <sip:againdevice@192.168.35.116:5060>
From: <sip:againdevice@192.168.35.222>;tag=1833694817
To: <sip:againdevice@192.168.35.116:5060>;tag=as7ca216c6
Call-ID: 88674963
CSeq: 20 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '88674963' Method: ACK
Really destroying SIP dialog '0a9455283166dcb47a86041118210bfa@127.0.0.1:5060' Method: OPTIONS

Is there any way to make call with this intercom?

Many thanks!

2

No. It is too badly broken.

Also please explain what you hope to achieve by using insecure=invite and by setting allowguest outside the general section. Also what you expect to achieve by setting host to 0.0.0.0?

Please note that directrtpsetup never reached supported status, and requires m lines.

3

Hello. Thanks for response.

insecure=invite override insecure=no in general setting. Without this asterisk send SIP/2.0 401 Unauthorized as response to invite packet.

You are allright, that allowguest must be in general section. It is mistake…

Setting host to 0.0.0.0 is some historical crazy thing, that I forget remove.

Thanks!

4

insecure=invite should only make a difference if there is a secret, and remotesecret is a better solution, in that case.

The problem that insecure=invite originally solved was that secret causes both way authentication, but ITSP’s generally don’t authenticate themselves, even though they require the caller to authenticate.

5

Hello,
There is very little source about the Hikvision DS-KV8102 SIP capabilities. So I would like to ask you about your case. Could you reach that your device registered itself to Asterisk?

I should call my device from an internal extension via Asterisk (v13.17.1) what is seems to me problematic without device registration on SIP server. The reverse way is working (calling from the intercom as guest a registered internal extension), but unfortunately that is not solve my problem.

Thank you in advance. Any help is appreciated.

6

clev22: Please start your own thread.