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

Sat Oct 5 02:49:45 MSD 2013

Do you have any logs from this?
I have not seen this before and I am not sure what would be causing that.
Full console log with sip traces.  POST on JIRA

On Thu, Oct 3, 2013 at 6:00 PM, sangdrax8 <sangdrax8 at gmail.com> wrote:

> 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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
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>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>

-- 
Anthony Minessale II

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