Bug-report: Unable to create local SDP at reinvite with SRTP

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Hi PJSIP-maintainers,
 
I have a reproducable problem with PJSUA at receiving a reinvite with 2 media-lines.
 
At a re-INVITE with new SDP PJSUA (release 1.8.5) gives next error:
 10:42:56.092   pjsua_call.c  Call 0: received updated media offer
 10:42:56.092   pjsua_call.c  Unable to create local SDP: Invalid SDP media transport protocol (PJMEDIA_SDP_EINPROTO) [status=220036]

Call scenario:
First INVITE has been answered succesfully with 200, RTP is flowing
Next at the re-INVITE with new remote SDP, no local SDP could be created.
 
The remote SDP contains 2 media-lines, first one with SRTP, next without:
v=0
o=- 21 21 IN IP4 192.168.9.42
s=Polycom IP Phone
c=IN IP4 192.168.9.42
t=0 0
m=audio 49212 RTP/SAVP 8 0 18 96
a=crypto:55 AES_CM_128_HMAC_SHA1_80 inline:yLw6kigvfpbr6T/aPxuTHZJLz4aj65xEf2Fo/0D/
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
m=audio 49212 RTP/AVP 8 0 18 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
 
The complete PJSUA-log is attached.
 
I have disabled SRTP in PJSUA by the following line in config_site.h :  #define PJMEDIA_HAS_SRTP 0
[Note: enabling SRTP by removing this line did not solve the problem, only gave another error 
 "Invalid SDP media transport for SRTP (PJMEDIA_SRTP_ESDPINTRANSPORT) [status=220227]"]
 
I found that with callstack:
  transport_udp.c - transport_encode_sdp
  transport.h - pjmedia_transport_encode_sdp
  pjsua_media.c - pjsua_media_channel_create_sdp
  pjsua_call.c - pjsua_call_on_rx_offer 
 
 ?transport_udp.c - transport_encode_sdp? returns PJMEDIA_SDP_EINPROTO
because it is called with media_index 0, and 
rem_sdp->media[media_index]->desc.transport 
contains ?RTP/SAVP? instead of ?RTP/AVP?.
 
 
Can you give me some guidance how to solve this problem?
 
 
Thank you,
 
Aire
 
 
 
  		 	   		  
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