Recently setup asterisk 18 and after setup everything getting below error.
ERROR[29929]: pjproject: <?>: sip_inv.c .Error parsing/validating SDP body: Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)
Not sure, where to check or anything is blocking?
What is the endpoint configuration? Someone else saw this, I believe, when they used “allow=all” which is highly discouraged.
I have configured PJSIP endpoint with below configurations.
[202]
type=endpoint
aors=202
auth=202-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
disallow=all
allow=ulaw,h264
context=internal
callerid=202 <202>
dtmf_mode=rfc4733
direct_media=yes
aggregate_mwi=yes
use_avpf=yes
rtcp_mux=yes
max_audio_streams=1
max_video_streams=1
bundle=yes
ice_support=yes
media_use_received_transport=yes
trust_id_inbound=yes
user_eq_phone=no
media_encryption=dtls
timers=yes
timers_min_se=90
media_encryption_optimistic=no
refer_blind_progress=yes
refer_blind_progress=yes
rtp_timeout=30
rtp_timeout_hold=300
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
dtls_rekey=0
dtls_auto_generate_cert=yes
webrtc=yes
You would need to provide a SIP trace, then. And you’re using a WebRTC endpoint?
Yes, using a WebRTC endpoint.
I’m not able to login with the ext. Directly getting this error on CLI and not getting any request on server side at the time of login.
Are you stating that it does not show up when “pjsip set logger on” is done? Is this an inbound call? Can you add debug to the WebRTC client to output the SDP? Is your WebRTC client modifying the SDP?
Are you stating that it does not show up when “pjsip set logger on” is done?
Yes,
Is your WebRTC client modifying the SDP?
No
<— Received SIP request (411 bytes) from UDP:193.29.14.125:5075 —>
OPTIONS sip:100@xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP 193.29.14.125:5075;branch=z9hG4bK-3410151004;rport
Content-Length: 0
From: "sipvicious"sip:100@1.1.1.1;tag=36303465336165333133633401373337313137393331
Accept: application/sdp
User-Agent: friendly-scanner
To: "sipvicious"sip:100@1.1.1.1
Contact: sip:100@193.29.14.125:5075
CSeq: 1 OPTIONS
Call-ID: 199333431516105535559074
Max-Forwards: 70
<— Transmitting SIP response (853 bytes) to UDP:193.29.14.125:5075 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.29.14.125:5075;rport=5075;received=193.29.14.125;branch=z9hG4bK-3410151004
Call-ID: 199333431516105535559074
From: “sipvicious” sip:100@1.1.1.1;tag=36303465336165333133633401373337313137393331
To: “sipvicious” sip:100@1.1.1.1;tag=z9hG4bK-3410151004
CSeq: 1 OPTIONS
Accept: application/dialog-info+xml, application/pidf+xml, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summary, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: text/plain
Accept-Language: en
Server: FPBX-15.0.17.12(18.1.1)
Content-Length: 0
That is an OPTIONS request presumably from someone trying to hack your system. If your system is open and not firewalled, then it is entirely possible that the PJMEDIA_SDP_EMISSINGRTPMAP message is a result of that - an improper call attempt from an attacker.
Solved it!!.
there was an issue from firewall which was blocking SIP request.
Thanks for your support and time.