[OpenSIPS-Users] all sip body headers regarding video removed

[Cindy Leung]

Hello,

I have to delete codec for similar reasons.  I'm trying to use codec_delete_except_re().

Here's the incoming SDP:
m=audio 49152 RTP/AVP 123 122 125 121 124 9 0 8 113 100
a=sendrecv
a=ptime:20
a=rtpmap:123 G7221/32000
a=fmtp:123 bitrate=48000
a=rtpmap:122 G7221/32000
a=fmtp:122 bitrate=32000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=32000
a=rtpmap:121 G7221/32000
a=fmtp:121 bitrate=24000
a=rtpmap:124 G7221/16000
a=fmtp:124 bitrate=24000
a=rtpmap:113 MP4A-LATM/32000
a=rtpmap:100 telephone-event/8000
m=video 49154 RTP/AVP 98 34 96 97
a=sendrecv
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=42801f; max-fs=21600
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=3;QCIF=2
a=rtpmap:96 H263-1998/90000
a=fmtp:96 CIF4=1;CIF=1;QCIF=1;F=1;I=1;J=1;T=1
a=rtpmap:97 H263-2000/90000
a=fmtp:97 profile=1; level=42
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
m=application 49158 RTP/AVP 110
a=sendrecv
a=rtpmap:110 H224/4800

... which then goes through 
codec_delete_except_re("PCMU|G729|telephone-event");

... and I end up getting
m=audio 28964 RTP/AVP 9 0 8 100
a=sendrecv
a=ptime:20
a=rtpmap:100 telephone-event/8000
m=video 28734 RTP/AVP 
a=sendrecv
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
m=application 27676 RTP/AVP 
a=sendrecv

Two issues I'm seeing here:
1.  Static audio payloads are not being removed (in this case G722 and G711a).  However, if the a= lines are provided, then they get removed as expected.
2.  The m= and a= lines for "video" and "application" are still there while the payload types are all deleted.  

Thanks.


Cinthia



On Aug 4, 2011, at 11:41 AM, Dani Popa wrote:

> Thanks,
> 
> I already did that.
> 
> Dani
> 
> On 08/04/11 18:19, Razvan Crainea wrote:
>> Hi Dani,
>> 
>> You can try by deleting the most common video codecs (like H261, H263, H264).
>> You can do that using the codec_delete[1] functions from the textops module.
>> I think you should also replace the video port with 0.
>> 
>> [1] http://www.opensips.org/html/docs/modules/devel/textops.html#id293910
>> 
>> Regards,
>> 
>> Razvan Crainea
>> OpenSIPS Developer
>> 
>> 
>> On 04.08.2011 18:03, Dani Popa wrote:
>>> Hi,
>>> 
>>> In fact, i have some problems with one of my pstn gw's that send "400 Incorrect content length", i think, because of too long sip packet. So, because it is pstn, i want to remove video capability(many lines in first invite packet).
>>> 
>>> Dani
>>> 
>>> On 08/04/11 17:02, Razvan Crainea wrote:
>>>> Hi Dani,
>>>> 
>>>> Why would you do that? If you don't want to allow video, you can simply replace the video port in the "m=" line with 0.
>>>> 
>>>> Regards,
>>>> 
>>>> Razvan Crainea
>>>> OpenSIPS Developer
>>>> 
>>>> 
>>>> On 04.08.2011 16:58, Dani Popa wrote:
>>>>> Hi all,
>>>>> 
>>>>> How can i remove all sip video body headers regardin video. Should i remove any line from body after "m=video", or how. Please give me a hint, if you have.
>>>>> 
>>>>> Thanks,
>>>>> Dani
>>>>> 
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