Call Tone dissapears

Asterisk Asterisk SIP
1

We have observed some strange behavior with a customer. A telephone calls a provider via a proxy and asterisk. The caller first hears a ringtone, which then disappears until the call is answered. Next, audio is successful in both directions.

It is noticeable that the provider initially signals 180 ringing. This then also goes as 180 ringing via the proxy to the telephone. This is why the caller hears the call sign.

After a few seconds an 183 ringing without SDP follows from the provider. Asterisk then sends an 183 with SDP to the phone via proxy. The phone then turns off the callsign, as this should now come from the provider.

The setting ignore_183_without_sdp has no influence on what happens. Although the provider (Deutsche Telekom) is large, this error only occurs with this customer (other customers either receive 183 with SDP or a direct 183 with SDP without previous 180 ringing).

Asterisk version is 18.19.0 with PJSIP

Exemplary call flow shortened to the essentials:

Time: 0 to provider:

INVITE sip:destination
Via: xxx
From: xxx
To: xxx
Supported: replaces, norefersub, histinfo
Max-Forwards: 70
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 2117669693 2117669693 IN IP4 93.241.79.172
s=Asterisk
c=IN IP4 93.241.79.172
t=0 0
m=audio 10044 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Time +5 sec from provider:

SIP/2.0 180 Ringing
Via: xxx
From: xxx
To: xxx
P-Early-Media: inactive
Content-Length: 0

TIme +5 Sec to the phone:

SIP/2.0 180 Ringing
Via: xxx
From: xxx
To: xxx
Content-Length:  0

Time +20 Sec from provider:

SIP/2.0 183 Session Progress
Via: xxx
From: xxx
To: xxx
P-Early-Media: inactive
Content-Length: 0

Time +20 Sec to the phone:

SIP/2.0 183 Session Progress
Via: xxx
From: xxx
To: xxx
Content-Type: application/sdp
Content-Length:   237

v=0
o=- 1725597608 1725597610 IN IP4 192.168.2.60
s=Asterisk
c=IN IP4 192.168.2.60
t=0 0
m=audio 10038 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Time +63 sec from provider (Audio in both directions)

SIP/2.0 200 OK
Via: xxx
From: xxx
To: xxx
Session-Expires: 1800;refresher=uas
Supported: histinfo
Supported: 100rel
Supported: timer
Content-Type: application/sdp
Content-Length: 269

v=0
o=ccs-0-615-2 06122898571627 1427108069 IN IP4 217.0.170.42
s=-
c=IN IP4 217.0.170.42
t=0 0
a=rtpengine:28f82c5516cd
m=audio 33418 RTP/AVP

Any ideas why this happens?

Thanks in advance
Karsten

2

Is the 180 with or with out SDP ?

But your 183 contains no SDP, so the phone is wrong.

You might want to consider adding a new-feature request in to the bug tracker. Something like “add PJSIP option to translate all 183s without SDPs to 180s without SDPs” might help work-a-round this.

3

Feature requests go here[1].

[1] GitHub - asterisk/asterisk-feature-requests: A place to submit feature and improvement requests for the Asterisk project. Contains no code.

4

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