PJSIP Incomplete SIP Message

Asterisk Asterisk Support
1

I have an Asterisk server running Asterisk certified-20.7-cert1. I’m attempting to send a FAX using the SendFax application via the T.38 protocol. While the SIP session and T.38 negotiation are successfully established, there is an issue when Asterisk attempts to switch back to audio media after the FAX transmission. Asterisk sends a re-INVITE with incomplete SDP content, as shown below:

INVITE sip:+12345@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP 5.6.7.8:5060;rport;branch=z9hG4bKPj48f9be62-8890-43b3-8e25-424b7dd8ea1c
From: "F1003092X" <sip:98765@5.6.7.8>;tag=beff0da7-fa0c-471c-a316-f8df78bb5fbe
To: <sip:12345@1.2.3.4>;tag=3930987644-294715371
Contact: <sip:98765@5.6.7.8:5060>
Call-ID: ce658d65-a6d5-4a20-8ed5-85b6f6f26266
CSeq: 16131 INVITE
Route: <sip:1.2.3.4;lr;r2=on;ftag=beff0da7-fa0c-471c-a316-f8df78bb5fbe>
Route: <sip:1.2.3.4:5060;lr;r2=on;ftag=beff0da7-fa0c-471c-a316-f8df78bb5fbe>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 30
User-Agent: MY-PBX
Content-Type: application/sdp
Content-Length:   117

v=0
o=MY 1071388670 1071388672 IN IP4 5.6.7.8
s=Asterisk
c=IN IP4 5.6.7.8
t=0 0
m=image 0 udptl t38

However, on another box using chan SIP on Asterisk 11.25.3, this issue does not occur. Here is an example of the successful re-INVITE message from the older system:

INVITE sip:+12345@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK728463a6;rport
Route: <sip:1.2.3.4;r2=on;lr=on;ftag=as1689a358>,<sip:1.2.3.4:5060;r2=on;lr=on;ftag=as1689a358>
Max-Forwards: 70
From: "F1003092X" <sip:98765@5.6.7.8>;tag=as1689a358
To: <sip:12345@ISP>;tag=3930987329-460612546
Contact: <sip:98765@5.6.7.8:5060>
Call-ID: 61f40f0477721be03d81f0383f0f1136@5.6.7.8:5060
CSeq: 103 INVITE
User-Agent: MY-PBX
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 249

v=0
o=SIP 637177864 637177866 IN IP4 5.6.7.8
s=Asterisk
c=IN IP4 5.6.7.8
t=0 0
m=audio 12372 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Here is the pjsip endpoint configuration currently in use:

[ISP]
 type=endpoint
 transport=transport-udp-nat
 context=MAIN
 acl=local
 disallow=all
 allow=alaw,ulaw
 aors=ISP
 direct_media=no
 identify_by=header,ip
 allow_subscribe=no
 t38_udptl=yes
 t38_udptl_ec=redundancy
 fax_detect=yes

Any idea why the re-INVITE SDP content appears incomplete in the Asterisk certified-20.7-cert1 setup and how to resolve this issue so that Asterisk properly switches back to audio media after sending the FAX via T.38?

2

There is nothing to do from a configuration perspective, and unless you are a support agreement customer issue reports will not be accepted against certified.

3

We initially attempted to resolve the issue using Asterisk version 20.9.1, but encountered the same problem. Hoping that the certified version might behave differently, we also tried Asterisk certified-20.7-cert1, but the issue persisted.

4

You can file an issue[1] with complete step by step reproduction instructions, but there is no guarantee on if or when it would get looked into.

[1] Issues · asterisk/asterisk · GitHub

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5

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