[Freeswitch-dev] Freeswitch rejecting streams when a=rtpmap:96 /0
Sam Russell
sam.h.russell at gmail.com
Fri Oct 4 02:03:47 MSD 2013
I think Anthony put a patch on HEAD for this - the RFC states that this is
acceptable (the a= line should be ignored when the m= line has no port),
but there's a regex in recent versions of freeswitch that gets upset when
it can't match anything in front of the /0
On Fri, Oct 4, 2013 at 2:54 AM, sangdrax8 <sangdrax8 at gmail.com> wrote:
> 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
> @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
> @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
>
>
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