[Freeswitch-dev] Freeswitch rejecting streams when a=rtpmap:96 /0
sangdrax8
sangdrax8 at gmail.com
Fri Oct 4 03:00:19 MSD 2013
Yes head does handle this with out hanging up, but it also inserts it now
when the client didn't. So I was wondering why it was adding this which
breaks comparability with all older versions of FreeSWITCH which don't
insert this.
On Oct 3, 2013 6:03 PM, "Sam Russell" <sam.h.russell at gmail.com> wrote:
> 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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>>
>>
>>
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>>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
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>
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