I couldn't find all the details for my previous example, but I did find a 
similar case where there are a couple of AMR packets sent to PBX that makes the 
PBX decline the call.

I have attached the signaling somewhat anonymized.



Call flow

[cid:image001.png@01D47D1D.5DBD2EF0]

BR/pj



-----Original Message-----
From: Paul Kyzivat [mailto:pkyzi...@alum.mit.edu]
Sent: den 15 november 2018 19:27
To: Sundbaum Per-Johan (Telenor Sverige AB); 
sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] RTP with wrong payload



On 11/15/18 12:02 PM, Sundbaum Per-Johan (Telenor Sverige AB) wrote:

> G.722 was offered in the initial INVITE to PBX, but was not accepted

> by PBX, in 200OK from PBX there were only G.711A

>

> SDP in INITIAL invite:

> SDP PDU

>    v=0

>    o=BroadWorks 400693062 1 IN IP4 195.54.102.188

>    s=-

>    c=IN IP4 195.54.102.188

>    t=0 0

>    m=audio 15148 RTP/AVP 8 110 111 0 96

>    b=AS:141

>    a=rtpmap:8 PCMA/8000

>    a=rtpmap:9 G722/8000

>    a=rtpmap:97 AMR/8000

>    a=fmtp:97 
> mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0

>    a=rtpmap:110 AMR/8000

>    a=fmtp:110 mode-change-period=2; mode-change-capability=2; 
> mode-change-neighbor=1; max-red=0

>    a=rtpmap:0 PCMU/8000

>    a=rtpmap:96 telephone-event/8000

>    a=fmtp:96 0-15

>    a=maxptime:20

>    a=ptime:20



The above seems a bit odd:



- why is there no rtpmap for 111?



- why is there an rtpmap for 97 that isn't mentioned in the m-line?



And then, the problem you are reporting is that the *offerer* is receiving (on 
port 15148) packets with pt=9?



                             Thanks,

                             Paul





> SDP in 200OK for INVITE from PBX

> SDP PDU

>    v=0

>    o=- 6613665318425236764 2 IN IP4 172.18.8.21

>    s=MX-ONE

>    c=IN IP4 172.18.8.32

>    t=0 0

>    m=audio 30838 RTP/AVP 8 101

>    a=rtpmap:8 PCMA/8000

>    a=rtpmap:101 telephone-event/8000

>    a=ptime:20

>    a=sqn:0

>    a=cdsc:1 image udptl t38

>    a=cpar:a=T38FaxVersion:0

>    a=cpar:a=T38MaxBitRate:14400

>    a=cpar:a=T38FaxRateManagement:transferredTCF

>    a=cpar:a=T38FaxMaxBuffer:9772

>    a=cpar:a=T38FaxMaxDatagram:1472

>    a=cpar:a=T38FaxUdpEC:t38UDPRedundancy

>    a=sendrecv

>

> BR/pj

>

> -----Original Message-----

> From: 
> sip-implementors-boun...@lists.cs.columbia.edu<mailto:sip-implementors-boun...@lists.cs.columbia.edu>

> [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of

> Paul Kyzivat

> Sent: den 15 november 2018 17:37

> To: 
> sip-implementors@lists.cs.columbia.edu<mailto:sip-implementors@lists.cs.columbia.edu>

> Subject: Re: [Sip-implementors] RTP with wrong payload

>

> On 11/15/18 1:21 AM, Sundbaum Per-Johan (Telenor Sverige AB) wrote:

>> I should have given more details, in the example I gave there was actual a 
>> couple of G.722 packets that was marked with payload type G.722 received in 
>> a session where G.711A(PCMA/8000) was established as the agreed codec, the 
>> receiving PBX did not have support for G.722.

>> As I interpret  RFC 3550 the PBX should drop the G.722 packets and let the 
>> session continue, and same applies also in case where G.722 is supported by 
>> PBX,  am I wrong ?

>

> Just to be sure...

>

> Are you saying that G.722 was not negotiated at all? Or that it wasn't the 
> first codec in the list?

>

> If multiple codecs are negotiated, then it is permissible to use them, and 
> even mix their use. (This is most often a cobmination of telephone-events 
> with another codec, but isn't limited to that.

>

> Can you post the actual offer/answer SDP that was used to negotiate the 
> session?

>

>                          Thanks,

>                          Paul

>

>> BR/pj

>>

>> -----Original Message-----

>> From: 
>> sip-implementors-boun...@lists.cs.columbia.edu<mailto:sip-implementors-boun...@lists.cs.columbia.edu>

>> [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of

>> Dale R. Worley

>> Sent: den 15 november 2018 05:10

>> To: Paul Heitkemper

>> Cc: 
>> sip-implementors@lists.cs.columbia.edu<mailto:sip-implementors@lists.cs.columbia.edu>

>> Subject: Re: [Sip-implementors] RTP with wrong payload

>>

>> Paul Heitkemper <pheitkem...@iedaudio.com<mailto:pheitkem...@iedaudio.com>> 
>> writes:

>>> RFC 3550 Section 5.1

>>>

>>> " A receiver MUST ignore packets with payload types that it does not

>>> understand."

>>

>> Though this rule is based on the payload type code, and not the encoding.  
>> The original post says only that the packets contain G.722 data, but if that 
>> data is marked with the payload type code that was negotiated for G.711A, 
>> the recipient will try to decode it as G.711A.

>> Perhaps the recipient can determine that the data is invalid (as G.711A) and 
>> discard it, but more likely it will decode it into some sort of noise which 
>> it will present to the user.

>>

>> Dale

>> _______________________________________________

>> Sip-implementors mailing list

>> Sip-implementors@lists.cs.columbia.edu<mailto:Sip-implementors@lists.cs.columbia.edu>

>> https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

>> _______________________________________________

>> Sip-implementors mailing list

>> Sip-implementors@lists.cs.columbia.edu<mailto:Sip-implementors@lists.cs.columbia.edu>

>> https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

>>

>

> _______________________________________________

> Sip-implementors mailing list

> Sip-implementors@lists.cs.columbia.edu<mailto:Sip-implementors@lists.cs.columbia.edu>

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>


Message #1
------------------------------------------------------
IP
------------------------------------------------------
   Source IP address = 195.54.102.102
   Destination IP address = 10.12.11.32
------------------------------------------------------
UDP
------------------------------------------------------
   Source port = 5060
   Destination port = 5060
------------------------------------------------------
SIP
------------------------------------------------------
   Request Line 
      INVITE sip:+467661...@sip.somepbx.se:5060 SIP/2.0
   Headers 
      Via: SIP/2.0/UDP 195.54.102.102:5060;branch=z9hG4bK3nr5k930387j6fobiun0.1
      To: ". ."<sip:+467661...@sip.somepbx.se>;cscf
      From: 
"+4670713753"<sip:+4670713...@pbx.telenor.se:37978;user=phone>;tag=1574694631-1542267460534-
      Call-ID: BW083740534151118-1959892700@10.49.24.132
      CSeq: 187100636 INVITE
      Max-Forwards: 68
      Content-Length: 508
      Contact: <sip:195.54.102.102:5060;transport=udp>
      Content-Type: application/sdp
      Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, 
UPDATE
      Accept: application/btbc-session-info
      Accept: application/media_control+xml
      Accept: application/sdp
      Accept: application/x-hotsip-filetransfer+xml
      Accept: multipart/mixed
      Supported:  timer
      Privacy: none
      Min-SE: 120
      Session-Expires: 1800
      Recv-Info: x-broadworks-client-session-info
      Route: <sip:10.12.11.32:5060;lr>
   Body 
      SDP PDU 
         v=0
         o=BroadWorks 693770312 1 IN IP4 195.54.102.102
         s=-
         c=IN IP4 195.54.102.102
         t=0 0
         m=audio 12002 RTP/AVP 8 110 111 0 96
         b=AS:141
         a=rtpmap:8 PCMA/8000
         a=rtpmap:110 AMR/8000
         a=fmtp:110 mode-change-period=2; mode-change-capability=2; 
mode-change-neighbor=1; max-red=0
         a=rtpmap:111 AMR/8000
         a=fmtp:111 octet-align=1; mode-change-period=2; 
mode-change-capability=2; mode-change-neighbor=1; max-red=0
         a=rtpmap:0 PCMU/8000
         a=rtpmap:96 telephone-event/8000
         a=fmtp:96 0-15
         a=maxptime:20
         a=ptime:20
 
Message #2
------------------------------------------------------
IP
------------------------------------------------------
   Source IP address = 10.12.11.32
   Destination IP address = 195.54.102.102
------------------------------------------------------
UDP
------------------------------------------------------
   Source port = 5060
   Destination port = 5060
------------------------------------------------------
SIP
------------------------------------------------------
   Status Line 
      SIP/2.0 100 Trying
   Headers 
      Via: SIP/2.0/UDP 
195.54.102.102:5060;rport=5060;received=195.54.102.102;branch=z9hG4bK3nr5k930387j6fobiun0.1
      Call-ID: BW083740534151118-1959892700@10.49.24.132
      From: "+4670713753" 
<sip:+4670713...@pbx.telenor.se;user=phone>;tag=1574694631-1542267460534-
      To: ". ." <sip:+467661...@sip.somepbx.se>;cscf
      CSeq: 187100636 INVITE
      
      Content-Length:  0
Message #3
------------------------------------------------------
IP
   Source IP address = 10.12.11.32
   Destination IP address = 195.54.102.102
------------------------------------------------------
UDP
------------------------------------------------------
   Source port = 5060
   Destination port = 5060
   Length Field = 893
   Checksum = 32cd'H
------------------------------------------------------
SIP
------------------------------------------------------
   Status Line 
      SIP/2.0 183 Session Progress
   Headers 
      Via: SIP/2.0/UDP 
195.54.102.102:5060;rport=5060;received=195.54.102.102;branch=z9hG4bK3nr5k930387j6fobiun0.1
      Call-ID: BW083740534151118-1959892700@10.49.24.132
      From: "+4670713753" 
<sip:+4670713...@pbx.telenor.se;user=phone>;tag=1574694631-1542267460534-
      To: ". ." 
<sip:+467661...@sip.somepbx.se>;tag=ce2341f5-aa0b-41ed-ac17-051485521bfd;cscf
      CSeq: 187100636 INVITE
      
      Contact: <sip:10.13.11.32:5060>
      Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, REGISTER, MESSAGE, REFER
      Content-Type: application/sdp
      Content-Length:   248
   Body 
      SDP PDU 
         v=0
         o=- 693770312 3 IN IP4 10.13.11.32
         s=Asterisk
         c=IN IP4 10.13.11.32
         t=0 0
         m=audio 14234 RTP/AVP 8 0 96
         a=rtpmap:8 PCMA/8000
         a=rtpmap:0 PCMU/8000
         a=rtpmap:96 telephone-event/8000
         a=fmtp:96 0-16
         a=ptime:20
         a=maxptime:150
         a=sendrecv
Message #4
------------------------------------------------------
IP
------------------------------------------------------
   Source IP address = 10.13.11.32
   Destination IP address = 195.54.102.102
------------------------------------------------------
UDP
------------------------------------------------------
   Source port = 5060
   Destination port = 5060
------------------------------------------------------
SIP
------------------------------------------------------
   Status Line 
      SIP/2.0 183 Session Progress
   Headers 
      Via: SIP/2.0/UDP 
195.54.102.102:5060;rport=5060;received=195.54.102.102;branch=z9hG4bK3nr5k930387j6fobiun0.1
      Call-ID: BW083740534151118-1959892700@10.49.24.132
      From: "+4670713753" 
<sip:+4670713...@pbx.telenor.se;user=phone>;tag=1574694631-1542267460534-
      To: ". ." 
<sip:+467661...@sip.somepbx.se>;tag=ce2341f5-aa0b-41ed-ac17-051485521bfd;cscf
      CSeq: 187100636 INVITE
      Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, REGISTER, MESSAGE, REFER
      Contact: <sip:10.13.11.32:5060>
      Content-Type: application/sdp
      Content-Length:   248
   Body 
      SDP PDU 
         v=0
         o=- 693770312 3 IN IP4 10.13.11.32
         s=Asterisk
         c=IN IP4 10.13.11.32
         t=0 0
         m=audio 14234 RTP/AVP 8 0 96
         a=rtpmap:8 PCMA/8000
         a=rtpmap:0 PCMU/8000
         a=rtpmap:96 telephone-event/8000
         a=fmtp:96 0-16
         a=ptime:20
         a=maxptime:150
         a=sendrecv
 
Message #5
------------------------------------------------------
IP
------------------------------------------------------
   Source IP address = 10.13.11.32
   Destination IP address = 195.54.102.102
------------------------------------------------------
UDP
------------------------------------------------------
   Source port = 5060
   Destination port = 5060
------------------------------------------------------
SIP
------------------------------------------------------
   Status Line 
      SIP/2.0 603 Decline
   Headers 
      Via: SIP/2.0/UDP 
195.54.102.102:5060;rport=5060;received=195.54.102.102;branch=z9hG4bK3nr5k930387j6fobiun0.1
      Call-ID: BW083740534151118-1959892700@10.49.24.132
      From: "+4670713753" 
<sip:+4670713...@pbx.telenor.se;user=phone>;tag=1574694631-1542267460534-
      To: ". ." 
<sip:+467661...@sip.somepbx.se>;tag=ce2341f5-aa0b-41ed-ac17-051485521bfd;cscf
      CSeq: 187100636 INVITE
      
      Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, REGISTER, MESSAGE, REFER
      Reason: Q.850;cause=16
      Content-Length:  0
Message #6
------------------------------------------------------
IP
   Source IP address = 195.54.102.102
   Destination IP address = 10.13.11.32
------------------------------------------------------
UDP
------------------------------------------------------
   Source port = 5060
   Destination port = 5060
------------------------------------------------------
SIP
------------------------------------------------------
   Request Line 
      ACK sip:+467661...@sip.somepbx.se:5060 SIP/2.0
   Headers 
      Via: SIP/2.0/UDP 195.54.102.102:5060;branch=z9hG4bK3nr5k930387j6fobiun0.1
      CSeq: 187100636 ACK
      To: ". 
."<sip:+467661...@sip.somepbx.se>;cscf;tag=ce2341f5-aa0b-41ed-ac17-051485521bfd
      From: 
"+4670713753"<sip:+4670713...@pbx.telenor.se:37978;user=phone>;tag=1574694631-1542267460534-
      Call-ID: BW083740534151118-1959892700@10.49.24.132
      Max-Forwards: 68
      Route: <sip:10.13.11.32:5060;lr>
      Content-Length: 0

 

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