Asterisk sending INVITE without an SDP

I am using Asterisk 18.2.2. When I call from one extension to another, the INVITE that Asterisk is sending to the called extension is not including any SDP at all:

<--- Transmitting SIP request (1156 bytes) to TCP:10.75.22.212:37334 --->
INVITE sip:my_sip_account@10.75.22.212:37334;transport=tcp;pn-type=firebase;app-id=[redacted];pn-tok=[redacted];pn-timeout=0;pn-silent=1 SIP/2.0
Via: SIP/2.0/TCP 10.75.22.8:5060;rport;branch=z9hG4bKPj1fa40ba5-8fec-4db3-bcf4-b7720f73678a;alias
From: "[redacted]" <sip:[redacted]@10.75.22.8>;tag=e9c5b70a-8444-45f6-ad2c-2f0a06a1ec80
To: <sip:my_sip_account@10.75.22.212;pn-type=firebase;app-id=[redacted];pn-tok=[redacted];pn-timeout=0;pn-silent=1>
Contact: <sip:asterisk@10.75.22.8:5060;transport=TCP>
Call-ID: 265bf877-b221-45e2-b7b9-33d4193bc027
CSeq: 29228 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.2.2
Content-Length:  0


<--- Received SIP response (566 bytes) from TCP:10.75.22.212:37334 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.75.22.8:5060;rport;branch=z9hG4bKPj1fa40ba5-8fec-4db3-bcf4-b7720f73678a;alias
From: "[redacted]" <sip:[redacted]@10.75.22.8>;tag=e9c5b70a-8444-45f6-ad2c-2f0a06a1ec80
To: <sip:my_sip_account@10.75.22.212;pn-type=firebase;app-id==[redacted];pn-tok==[redacted];pn-timeout=0;pn-silent=1>
Call-ID: 265bf877-b221-45e2-b7b9-33d4193bc027
CSeq: 29228 INVITE
Content-Length: 0

The call ends up being terminated immediately upon answer by the caller with:

[May 20 05:31:00] ERROR[4901]: res_pjsip_session.c:902 handle_incoming_sdp:  PJSIP/my_sip_account-00000012: Couldn't negotiate stream 0:audio-0:audio:sendrecv (ulaw|opus|codec2|g723|alaw|gsm|g726|g726aal2|adpcm|slin|slin12|slin16|slin24|slin32|slin44|slin48|slin96|slin192|lpc10|g729|speex|speex16|speex32|ilbc|g722|siren7|siren14|testlaw|g719|silk8|silk12|silk16|silk24)   

Ultimately the above error and the call terminating are the symptom I am trying to resolve, but the INVITE without an SDP seems suspicious. Is it related?

This typically happens when the configuration, which you didn’t provide, allows all codecs.

Wait. Allowing all codecs causes a call to fail? Why would that happen?

But it seems you are correct. Just choosing a codec (more or less at random) and disallowing it seems to have resolved the issue.

That surely is a bug, yes?

https://issues.asterisk.org/jira/browse/ASTERISK-29185

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