[Freeswitch-dev] Freeswitch rejecting streams when a=rtpmap:96 /0

[sangdrax8]

It seems that the current head has actually started producing this "rtpmap:
96 /0", when my client devices do not include it in the sdp.  This causes
problems when I am trying to upgrade one switch at a time in my production
environment.  I assume (will test here shortly) that if all switches are on
the newer version, this wouldn't be an issue as the indicated patch should
handle it.

I would really like to be able to upgrade my boxes individually, but if an
upgraded box add's this to the SDP, my non-upgraded boxes will kill the
call.

The following is an example SDP trace from my latest head (10.01.2013)
showing the SDP from the client device is normal, and then what is sent to
the other freeswitch (my currently non upgraded one) includes this problem
causing rtpmap.  I should note that this seems to only be the case when I
activate zrtp on the clients, which causes my dial plan to set
proxy_media=true.

I haven't opened a bug in jira, I wanted to confirm if this rtpmap is in
fact a bug.  If there is a good reason it is there now and isn't in my
older freeswitch versions, then I may have to attempt to upgrade all at one
time.

recv 1072 bytes from tls/[1.1.1.1]:50272 at 13:42:16.509540:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK5N858D7BX46je
   Contact: <sip:15715550001 at 192.168.10.136:60068;transport=tls>
   From: "Test 1" <sip:17035550001 at securevoice>;tag=jSmQZ67etF8pa
   Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a
   CSeq: 50067799 INVITE
   To: <sip:15715550001 at 192.168.10.136:60068
;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB
   Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE
   Supported: replaces, path
   Content-Type: application/sdp
   Content-Length: 534

   v=0
   o=- 98627 19401 IN IP4 192.168.10.136
   s=menduco
   c=IN IP4 192.168.10.136
   t=0 0
   m=audio 58620 RTP/SAVP 9 101
   a=rtpmap:101 telephone-event/8000
   a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi
   a=ptime:30
   a=fmtp:101 0-15
   a=zrtp-hash:1.10
99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0
   a=sendrecv
   m=audio 0 RTP/AVP 9 0 101
   a=rtpmap:101 TELEPHONE-EVENT/8000
   a=ptime:30
   a=fmtp:101 0-15
   a=zrtp-hash:1.10
18489BD8B5526F4C92AB6EB2A04022C25E8CB5D7E58D4E633FC4545783447F8C
   ------------------------------------------------------------------------
send 472 bytes to tls/[1.1.1.1]:50272 at 13:42:16.510594:
   ------------------------------------------------------------------------
   ACK sip:15715550001 at 192.168.10.136:60068;transport=tls SIP/2.0
   Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK6y1ya9QFtDX5S
   Max-Forwards: 70
   From: "Test 1" <sip:17035550001 at securevoice>;tag=jSmQZ67etF8pa
   To: <sip:15715550001 at 192.168.10.136:60068
;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB
   Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a
   CSeq: 50067799 ACK
   Contact: <sip:mod_sofia at 2.2.2.2:5061;transport=tls>
   Content-Length: 0
------------------------------------------------------------------------
send 1516 bytes to tls/[10.110.11.31]:47793 at 13:42:16.516187:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 10.110.11.31:5071
;branch=z9hG4bKD97QXp3XZgX4a;rport=47793
   From: "Test 1" <sip:17035550001 at 10.110.11.31>;tag=DU0S42e40FNpg
   To: <sip:15715550001 at dcprotopop1.private.sec
:5071;transport=tls>;tag=tcXjHQ6gme91S
   Call-ID: 6fd5bd89-a6d4-1231-8291-005056ac002c
   CSeq: 50067799 INVITE
   Contact: <sip:15715550001 at 10.110.11.32:5071;transport=tls>
   User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131001T160636Z~6d2280df08
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 408
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:15715550001 at dcprotopop1.private.sec
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Outbound Call" <sip:15715550001 at dcprotopop1.private.sec
>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1167543025 1167543026 IN IP4 10.110.11.32
   s=FreeSWITCH
   c=IN IP4 10.110.11.32
   t=0 0
   m=audio 27868 RTP/SAVP 9 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi
   a=ptime:30
   a=zrtp-hash:1.10
99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0
   m=audio 0 RTP/AVP 9 0 96
   a=rtpmap:96 /0
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20131003/b0f7b568/attachment.html