Calling station provides 8 video codecs variants in rtpmap, 4 of these is one allowed codec - H264. But asterisk rewrites SDP and have only one variant in outgoing offer.
Client sends
....
m=video 39617 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 122 127 121 125 107 108 109 35 36 120 119 124
c=IN IP4 95.29.193.106
a=rtpmap:96 VP8/90000
a=rtpmap:97 rtx/90000
a=rtpmap:98 VP9/90000
a=rtpmap:99 rtx/90000
a=rtpmap:100 VP9/90000
a=rtpmap:101 rtx/90000
a=rtpmap:102 H264/90000
a=rtpmap:122 rtx/90000
a=rtpmap:127 H264/90000
a=rtpmap:121 rtx/90000
a=rtpmap:125 H264/90000
a=rtpmap:107 rtx/90000
a=rtpmap:108 H264/90000
a=rtpmap:109 rtx/90000
a=rtpmap:35 AV1/90000
a=rtpmap:36 rtx/90000
a=rtpmap:120 red/90000
a=rtpmap:119 rtx/90000
a=rtpmap:124 ulpfec/90000
a=fmtp:97 apt=96
a=fmtp:98 profile-id=0
a=fmtp:99 apt=98
a=fmtp:100 profile-id=2
a=fmtp:101 apt=100
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=fmtp:122 apt=102
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=fmtp:121 apt=127
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=fmtp:107 apt=125
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=fmtp:109 apt=108
a=fmtp:36 apt=35
a=fmtp:119 apt=120
a=rtcp:9 IN IP4 0.0.0.0
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-c
Asterisk rewrite it as
m=video 19886 UDP/TLS/RTP/SAVPF 99
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 A5:6D:49:B2:B4:C6:B1:35:BD:7E:81:08:4A:0F:16:68:B1:AB:02:10:B5:E8:75:4C:8B:E2:97:6A:21:E9:F5:ED
a=ice-ufrag:04b8d0860b0abcb130a16ce30bba4a17
a=ice-pwd:39e7c82d6c00c6032a02834467853f46
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=42001F
a=sendrecv
a=rtcp-mux
a=ssrc:1574497547 cname:ac6f29f3-12df-4822-8892-51fd4cf8022e
a=msid:35743a7f-b0b7-41c3-86cc-96d8f3f6919f 93d3d702-0c8f-4465-a2ff-21a172ee2184
a=rtcp-fb:* transport-cc
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=rtcp-fb:* nack
a=extmap:1 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=mid:video-1
Called client want profile-level-id=42e01f and not support profile-level-id=42001f
Current endpoint configuration is
ParameterName : ParameterValue
========================================================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (h264|alaw)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : 385991
asymmetric_rtp_codec : true
auth : 385991
bind_rtp_to_media_address : false
bundle : true
call_group :
callerid : <unknown>
callerid_privacy : allowed_not_screened
callerid_tag :
codec_prefs_incoming_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_incoming_offer : prefer:configured, operation:intersect, keep:all, transcode:prevent
codec_prefs_outgoing_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_offer : prefer:configured, operation:union, keep:all, transcode:prevent
connected_line_method : invite
contact_acl :
context : intercom
cos_audio : 0
cos_video : 0
device_state_busy_at : 0
direct_media : false
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : true
dtls_auto_generate_cert : Yes
dtls_ca_file :
dtls_ca_path :
dtls_cert_file :
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key :
dtls_rekey : 0
dtls_setup : actpass
dtls_verify : Yes
dtmf_mode : rfc4733
fax_detect : false
fax_detect_timeout : 0
follow_early_media_fork : true
force_avp : false
force_rport : true
from_domain : ds.domofon-sip.ru
from_user :
g726_non_standard : false
ice_support : true
identify_by : username,auth_username
ignore_183_without_sdp : false
inband_progress : false
incoming_call_offer_pref : local
incoming_mwi_mailbox :
language : ru
mailboxes :
max_audio_streams : 1
max_video_streams : 10
media_address :
media_encryption : dtls
media_encryption_optimistic : false
media_use_received_transport : true
message_context : messages
moh_passthrough : false
moh_suggest : default
mwi_from_user :
mwi_subscribe_replaces_unsolicited : no
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : false
outbound_auth :
outbound_proxy :
outgoing_call_offer_pref : remote_merge
pickup_group :
preferred_codec_only : false
record_off_feature : automixmon
record_on_feature : automixmon
refer_blind_progress : true
rewrite_contact : true
rpid_immediate : false
rtcp_mux : true
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 15
rtp_symmetric : false
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_connected_line : yes
send_diversion : true
send_history_info : false
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
stir_shaken : false
sub_min_expiry : 0
subscribe_context : subscribe
suppress_q850_reason_headers : false
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport :
trust_connected_line : yes
trust_id_inbound : false
trust_id_outbound : false
use_avpf : true
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : yes
How to add streams with different fmtps ??