[asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127

Olle E Johansson reviewboard at asterisk.org
Fri Dec 19 14:39:54 CST 2014


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You may want to consider fixing RFC 5761 restrictions while you are in there... This prepares for RTP/RTCP muxing in WebRTC

"Given these constraints, it is RECOMMENDED to follow the guidelines
   in the RTP/AVP profile [7] for the choice of RTP payload type values,
   with the additional restriction that payload type values in the range
   64-95 MUST NOT be used.  Specifically, dynamic RTP payload types
   SHOULD be chosen in the range 96-127 where possible.  Values below 64
   MAY be used if that is insufficient, in which case it is RECOMMENDED
   that payload type numbers that are not statically assigned by [7] be
   used first."


- Olle E Johansson


On Dec. 19, 2014, 9:24 p.m., Scott Griepentrog wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4286/
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> (Updated Dec. 19, 2014, 9:24 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-24367
>     https://issues.asterisk.org/jira/browse/ASTERISK-24367
> 
> 
> Repository: Asterisk
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> Description
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> 
> Valid payload type codes are between 0 and 127 to allow for being stored in 7 bits.  During call setup, pjsip validates the SDP and will assert if it encounters an invalid payload type code (see pjmedia_sdp_validate() in pjmedia/src/pjmedia/sdp.c).  This assert will be hit if a call is placed to a pjsip endpoint with allow=all set.
> 
> To avoid this, the previous use 128 for the slin192 format has been changed to 95.
> 
> 
> Diffs
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>   /trunk/main/rtp_engine.c 429845 
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> Diff: https://reviewboard.asterisk.org/r/4286/diff/
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> 
> Testing
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> Tested with pjsip calls to allow=all configured extensions.
> 
> 
> Thanks,
> 
> Scott Griepentrog
> 
>

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