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Re: [Sip] Question regarding SDP 'M' Line
This probably should have been posted to sip-implementors, and certainly
not cross-posted to both sip and sipping. I'm replying here only because
there has been one reply here.
I can't remember where right now, but somewhere there has been a
clarification to the point that Guangwu raises. I believe it is now
considered ok to reuse an existing and active m-line for a different
media type.
Its illogical not to allow that. Since you could do it by first doing an
o/a to set the port to zero, and then another to reuse the line with the
zero port, why waste the bandwidth and processing to exchange those
meaningless messages?
Paul
Guangwu.He wrote:
> see my comments inline.
>
> Thanks & Regards,
>
> Guangwu.He
>
>
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> *From:* sip-bounces at ietf.org [mailto:sip-bounces at ietf.org] *On
> Behalf Of *Jayantheesh Srimushnam
> *Sent:* Wednesday, April 16, 2008 5:58 PM
> *To:* sip at ietf.org
> *Subject:* [Sip] Question regarding SDP 'M' Line
>
> Hi All,
>
>
>
> While adding a new media stream is it MUST to add a new ‘M’ line in
> the SDP or existing ‘M’ line can be replaced with the new one.
>
> Please go through the below specified example and clarify me which
> is the correct way of implementing?
>
>
>
> Example 1:
>
> ========
>
>
>
> Initial INVITE:
>
>
>
> m=audio 7100 RTP/AVP 0 101
>
> a=rtcp:7101
>
> a=sendonly
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
>
>
> Re-Invite:
>
>
>
> m=image 20112 udptl t38
>
> a=T38FaxVersion:0
>
> a=T38MaxBitRate:1440
>
>
>
> *[Guangwu]:the example is wrong;RFC3264 8.1*
>
> *New media streams are created by new additional media descriptions
> below the existing ones, or by reusing the "slot" used by an old
> media stream which had been disabled by setting its port to zero.
> *
>
> *in you example 1,the port is not zero;*
>
> **
>
> Example 2:
>
> ========
>
>
>
> Initial INVITE:
>
>
>
> m=audio 7100 RTP/AVP 0 101
>
> a=rtcp:7101
>
> a=sendonly
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
>
>
> Re-Invite:
>
>
>
> m=audio 7100 RTP/AVP 0 101
>
> a=rtcp:7101
>
> a=sendonly
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> m=image 5080 udptl t38
>
> c=IN IP4 10.101.210.30
>
> a=T38FaxVersion:0* *
>
>
>
> Thanks,
>
> Jayantheesh
>
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_______________________________________________
Sip mailing list https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors at cs.columbia.edu for questions on current sip
Use sipping at ietf.org for new developments on the application of sip