Please help me configure pjsip.conf to get the following items in the SDP of a h264 video call:
a=rtcp-fb:* ccm tmmbr
a=rtcp-fb:* ccm fir
a=rtcp-fb:* nack pli
a=rtcp-fb:* nack
The company I want to send the video calls to assures me they can send this SDP with an Asterisk server in a non-webrtc call. The issue is the video freezes too often. The provider tells me if I want to fix it I must enable the above rtcp featues. Based on my research, and setting webrtc=yes the best I can make appear in the SDP is the following:
rtcp-fb:* trasnsport-cc
rtcp-fb:* ccm fir
rtcp-fb:* goog-remb
rtcp-fb:* nack
Not exactly the same SDP, although it should be okay?. Which brings my second question: How can I additionally activate “a=rtcp-fb:* ccm tmmbr” and “a=rtcp-fb:* nack pli” on non-webrtc endpoints?. I get that this rtcp features are only available on webrtc calls (according to what I have read), but it is a current discussion I am having with the other engineers who claim they are currently enabling these rtcp features using non-webrtc Asterisk (albeit they won’t show me lol). Specifically when I set use_avpf=yes it seems to override my media_encrytion=no setting, since they claim they are receiving my call encrypted.
Can someone please confirm or correct me?
The following is my endpoint configuration. Please note it is on a Realtime database, therefore the NULL values. I assume anything NULL will be default. Again, this is suppposed to be a non-webrtc configuration.
Thanks
Configuration:
id = ‘endpoint_name’
allow = ‘h264,h263,vp8,alaw,ulaw,gsm’
asymmetric_rtp_codec = ‘no’
aors = ‘aor_name’
auth = NULL
bind_rtp_to_media_address = NULL
call_group = NULL
cos_audio = NULL
cos_video = NULL
context = ‘phone1’
connected_line_method = NULL
contact_user = ‘5161234567’
contact_deny = NULL
contact_permit = NULL
contact_acl = NULL
deny = NULL
device_state_busy_at = 1
direct_media = ‘no’
direct_media_method = NULL
direct_media_glare_mitigation = NULL
disable_direct_media_on_nat = NULL
dtmf_mode = NULL
dtls_verify = ‘no’
dtls_rekey = NULL
dtls_cert_file = NULL
dtls_private_key = NULL
dtls_cipher = NULL
dtls_ca_file = NULL
dtls_ca_path = NULL
dtls_setup = NULL
external_media_address = NULL
force_rport = ‘yes’
force_avp = ‘yes’
ice_support = ‘no’
inband_progress = NULL
identify_by = NULL
max_audio_streams = 50
max_video_streams = 50
media_address = NULL
media_encryption = ‘no’
media_encryption_optimistic = NULL
media_use_received_transport = ‘yes’
message_context = NULL
named_call_group = NULL
named_pickup_group = NULL
outbound_auth = NULL
outbound_proxy = NULL
permit = NULL
redirect_method = NULL
rewrite_contact = ‘yes’
rpid_immediate = NULL
rtcp_mux = ‘yes’
rtp_engine = NULL
rtp_ipv6 = NULL
rtp_keepalive = NULL
rtp_symmetric = NULL
rtp_timeout = NULL
rtp_timeout_hold = NULL
send_diversion = NULL
send_pai = NULL
send_rpid = NULL
sdp_owner = NULL
sdp_session = NULL
srtp_tag_32 = NULL
sub_min_expiry = NULL
subscribe_context = NULL
timers = NULL
timers_sess_expires = NULL
timers_min_se = NULL
tone_zone = NULL
tos_audio = NULL
tos_video = NULL
transport ‘tcp-transport’
trust_id_inbound = NULL
trust_id_outbound = ‘yes’
use_ptime = NULL
use_avpf = ‘yes’
webrtc = NULL