Incorrect SDP generated when answering call with RTP/AVP profile and when rtcp-fb is present
When answering incoming call with RTP/AVP profile pjsip generates wrong
sdp if at least one pjmedia_rtcp_fb_cap set to the account.
200/OK sdp from pjsip contains these lines:
m=audio 0 RTP/AVP 0 96
c=IN IP4 127.0.0.1
When receiving such sdp remote server sends BYE immediately.
Invite with RTP/SAVPF profile works with these settings.
When pjmedia_rtcp_fb_cap not set pjsip correctly answers to the invite
with RTP/AVP profile.
============
Account settings used (pjsip 2.9, TLS and UDP transport tested):
acc_cfg.mediaConfig.rtcpMuxEnabled = true
acc_cfg.natConfig.iceEnabled = true;
acc_cfg.mediaConfig.srtpOpt.keyings.push_back(PJMEDIA_SRTP_KEYING_DTLS_SRTP);
acc_cfg.mediaConfig.srtpOpt.keyings.push_back(PJMEDIA_SRTP_KEYING_SDES);
acc_cfg.mediaConfig.srtpUse = PJMEDIA_SRTP_OPTIONAL;
acc_cfg.mediaConfig.srtpSecureSignaling = 0;
RtcpFbCap cap;
cap.type = PJMEDIA_RTCP_FB_NACK;
cap.codecId = "H264";
cap.param = "pli";
acc_cfg.mediaConfig.rtcpFbConfig.caps.push_back(cap);
Incoming invite:
INVITE sip:1000@186.17.67.204:5190 SIP/2.0
Via: SIP/2.0/UDP 186.17.67.175:5160;rport;branch=z9hG4bK8cZ5XgUSK3UZH
Max-Forwards: 7
From: "l" sip:19570488@186.17.67.175;tag=HS4X211rv9c8D
To: sip:1000@186.17.67.204:5190
Call-ID: 4128449e-28f6-1238-2092-1cc1dee7574c
CSeq: 7446652 INVITE
Contact: sip:mod_sofia@186.17.67.175:5160
User-Agent: FreeSWITCH-mod_sofia/1.8.7~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, re
r
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 222
X-FS-Support: update_display,send_info
v=0
o=FreeSWITCH 1563957786 1563957787 IN IP4 186.17.67.175
s=FreeSWITCH
c=IN IP4 186.17.67.175
t=0 0
m=audio 43102 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
OK from pjsip:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
186.17.67.175:5160;rport=5160;branch=z9hG4bK8cZ5XgUSK3UZH
From: "l" sip:19570488@186.17.67.175;tag=HS4X211rv9c8D
To: sip:1000@186.17.67.204:5190;tag=X680y47rK3pyF
Call-ID: 4128449e-28f6-1238-2092-1cc1dee7574c
CSeq: 7446652 INVITE
Contact: sip:1000@186.17.67.204:5190;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.8.7~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 136
X-FS-Display-Name: Outbound Call
X-FS-Display-Number: sip:1000@186.17.67.204
X-FS-Support: update_display,send_info
v=0
o=- 3772989711 3772989713 IN IP4 186.17.67.146
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 0 RTP/AVP 0 96
c=IN IP4 127.0.0.1
Log does not contain anything interesting:
pjsua_call.c !Answering call 0: code=200
inv0x9b8b2864 ..SDP negotiation done: Success
pjsua_media.c ...Call 0: updating media..
pjsua_media.c .....Media stream call00:0 is destroyed
srtp0x9c4cec00 ....SRTP not active
pjsua_aud.c ....Audio channel update..
strm0x99bfe014 .....VAD temporarily disabled
strm0x99bfe014 .....Encoder stream started
strm0x99bfe014 .....Decoder stream started
pjsua_media.c ....Audio updated, stream #0: PCMU (sendrecv)
virtual void _PjCall::onCallMediaState(pj::OnCallMediaStateParam &)
pjsua_core.c ....TX 809 bytes Response msg 200/INVITE/cseq=7446662
(tdta0x9b8b8064) to UDP 186.17.67.204:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
186.17.67.204;rport=5060;received=186.17.67.204;branch=z9hG4bK3pyUU49j1NXrm
Call-ID: 4cb2c111-28f6-1238-aca7-826d5e9fa7cf
From: "l" sip:19570488@186.17.67.204;tag=Spy1Fpc5XctHD
To: sip:1000@186.17.67.146;ob;tag=1b6672c0-06b6-40a6-af5c-2e493e8a4a59
CSeq: 7446662 INVITE
Contact: sip:1000@186.17.67.146:54391;ob;+sip.ice
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 137
v=0
o=- 3772989711 3772989713 IN IP4 186.17.67.146
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 0 RTP/AVP 0 101
c=IN IP4 127.0.0.1
--end msg--