H.17 Media use cases

34.229-13GPPInternet Protocol (IP) multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP)Part 1: Protocol conformance specificationRelease 16TSUser Equipment (UE) conformance specification

H.17.1 Originating Voice, add video remove video / Fixed Broadband Access

H.17.1.1 Definition

Test to verify that the UE is able to add a bidirectional video component to an ongoing UE Originating IMS Multimedia telephony voice call for Fixed Broadband Access. This process is described in TS 24.229 [10], TS 24.173 [65] and TS 26.114 [66].

H.17.1.2 Conformance requirement

Same as described in 17.1.2.

H.17.1.3 Test purpose

1) To verify that when adding a video component to an ongoing IMS Multimedia Telephony voice call the UE performs correct exchange of SIP protocol signalling messages; and

2) To verify that within SIP signalling the UE performs correct SDP offer/answer exchanges for negotiating media and indicating preconditions for resource reservation (as described by TS 24.229 [10], clause 6.1); and

3) To verify that when removing the video component from the IMS Multimedia Telephony call the UE performs correct exchange of SIP and SDP protocol messages.

H.17.1.4 Method of test

Initial conditions

UE is configured with the home domain name, public and private user identities and SIP Digest Credentials.

SS is configured with the home domain name, public and private user identities and SIP Digest Credentials. SS is listening to SIP default port 5060 for both UDP and TCP protocols. SS is able to perform MD5 authentication algorithm for that IMPI, according to TS 33.203 [14] clause 6.1 and RFC 3310 [17]. SS has performed MD5 authentication with the UE and accepted the registration. SS has also performed the Originating voice Call according to C.21c.

Expected sequence

Step

Direction

Message

Comment

UE

SS

1

Make the UE attempt add IMS video to the voice call.

2

->

INVITE

UE sends re-INVITE with a SDP offer containing media lines for both voice and video

3

<-

100 Trying

SS sends a 100 Trying provisional response

4-6

Void

7

<-

200 OK

SS responds INVITE with 200 OK

8

->

ACK

UE acknowledges

9

Make UE release video from media call

10

->

INVITE

UE sends re-INVITE with a SDP offer indicating that the video component is removed from the call

11

<-

100 Trying

The SS responds with a 100 Trying provisional response

12

<-

200 OK

The SS responds re-INVITE with 200 OK

13

->

ACK

The UE acknowledges the receipt of 200 OK for re-INVITE

14

->

BYE

The UE releases the call with BYE

15

<-

200 OK

The SS sends 200 OK for BYE

NOTE: The default messages contents in annex A are used with condition "SIP Digest without TLS for Fixed Broadband Access" when applicable.

Specific Message Contents

INVITE (Step 2)

Use the default message "INVITE for MO Call" in clause A.2.1 with condition A5 (re-INVITE within a dialog) and the following exceptions:

Header/param

Value/Remark

Message-body

The following SDP types and values.

Session description:

– v=0

– o=(username) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE)

– s=(session name)

– c=IN (addrtype) (connection-address for UE) [Note 1]

– b=AS: (bandwidth-value)

Time description:

– t= (start-time) (stop-time)

Media description:

– m=audio (transport port) RTP/AVP (fmt)

– c=IN (addrtype) (connection-address for UE) [Note 1]

– b=AS: (bandwidth-value)

– b=RS: (bandwidth-value)

– b=RR: (bandwidth-value)

Attributes for media:

– a=rtpmap: (payload type) AMR/8000 [Note 6]

– a=fmtp: (format) mode-change-capability=2; max-red= (att-field) [Note 7]

– a=fmtp: (format)

– a=ptime:20

– a=maxptime:240

Media description:

– m=video (transport port) RTP/AVPF (fmt) or RTP/AVP (fmt) [Note 8]

– c=IN (addrtype) (connection-address for UE) [Note 1]

– b=AS: (bandwidth-value)

– b=RS: (bandwidth-value)

– b=RR: (bandwidth-value)

Attributes for media:

– a=tcap:1 RTP/AVPF [Note 8]

– a=pcfg:1 t=1 [Note 2]

– a=rtpmap: (payload type) H264/90000

– a=fmtp: (format) profile-level-id= (att-field)

Note 1: At least one "c=" field shall be present.

Note 2: Void.

Note 3: Void.

Note 4: a rate may be added to the “telephone-event” separated by “/” (e.g. “telephone-event/8000”)

Note 5: Void.

Note 6: The AMR channel number shall be “/1” or omitted.

Note 7: values from 0 to 220 are allowed

Note 8: The tcap/pcfg attributes are present if RTP/AVP is present on the m line.

200 OK (Step 7)

Use the default message “200 OK for other requests than REGISTER or SUBSCRIBE” in annex A.3.1 with the following exceptions:

Header/param

Value/Remark

Content-Type

media-type

application/sdp

Message-body

The following SDP types and values.

Session description:

– v=0

– o=- 1111111111 1111111111 IN (addrtype) (unicast-address for SS)

s=-

– c=IN (addrtype) (connection-address for SS)

– b=AS:30

Time description:

– t=0 0

Media description:

– m=audio (transport port) RTP/AVP (fmt) [Note 1, 4]

– b=AS: bandwidth-value) [Note 1]

– b=RS: (bandwidth-value) [Note 1]

– b=RR: (bandwidth-value) [Note 1]

Attributes for media:

– a=rtpmap: (payload type) AMR/8000/1 [Note 1]

– a=fmtp: (format) mode-change-capability=2; max-red=220 [Note 1]

– a=ptime:20

– a=maxptime:240

Media description:

– m=video (transport port) RTP/AVPF (fmt) or RTP/AVP (fmt) [Note 1]

– b=AS: (bandwidth-value)

– b=RS: (bandwidth-value)

– b=RR: (bandwidth-value)

Attributes for media:

– a= acfg:1 t=1 [Note 2]

– a=rtpmap: (payload type) H264/90000

– a=fmtp: (format) profile-level-id= (att-field)

Note 1: The value for fmt, bandwidth, payload type and format specific parameters copied from step 2.

Note 2: Present if tcap/pcfg attributes were included in step 2.

Note 3: Void.

Note 4: transport port is the port number of the SS (see RFC 3264 clause 6).

Note 5: Void.

INVITE (Step 10)

Use the default message "INVITE for MO Call" in clause A.2.1 with condition A5 (re-INVITE within a dialog) and the following exceptions:

Header/param

Value/Remark

Message-body

The following SDP types and values.

Session description:

– v=0

– o=(username) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE)

– s=(session name)

– c=IN (addrtype) (connection-address for UE) [Note 1]

– b=AS: (bandwidth-value)

Time description:

– t= (start-time) (stop-time)

Media description:

– m=audio (transport port) RTP/AVP (fmt)

– c=IN (addrtype) (connection-address for UE) [Note 1]

– b=AS: (bandwidth-value)

– b=RS: (bandwidth-value)

– b=RR: (bandwidth-value)

Attributes for media:

– a=rtpmap: (payload type) AMR/8000 [Note 6]

– a=fmtp: (format)

– a=ptime:20

– a=maxptime:240

Media description:

– m=video 0 RTP/AVPF (fmt)

– c=IN (addrtype) (connection-address for UE) [Note 1]

Note 1: At least one "c=" field shall be present.

Note 2: Void.

Note 3: Void.

Note 4: Void

Note 5: Void.

Note 6: The AMR channel number shall be “/1” or omitted.

Note 7: Void.

200 OK (Step 12)

Use the default message "200 OK for other requests than REGISTER or SUBSCRIBE" in annex A.3.1 with the following exceptions:

Header/param

Value/remark

Content-Type

media-type

application/sdp

Content-Length

Value

length of message-body

Message-body

SDP body of the 200 response copied from the received INVITE and modified as follows:

– "o=" line identical to previous SDP sent by SS except that sess-version is incremented by one

– IP address on "c=" line and, for audio, transport port on "m=" line changed to indicate to which IP address and port the UE should start sending the media;

H.17.1.5 Test requirements

The UE shall send requests and responses as described in clause H.17.1.4

H.17.2 Terminating Voice, add video remove video / Fixed Broadband Access

H.17.2.1 Definition

Test to verify that the UE is able to add a bidirectional video component to an ongoing UE terminating IMS Multimedia telephony voice call for Fixed Broadband Access. This process is described in TS 24.229 [10], TS 24.173 [65] and TS 26.114 [66].

H.17.2.2 Conformance requirement

Same as described in clause 17.2.2.

H.17.2.3 Test purpose

1) To verify that when adding a video component to an ongoing IMS Multimedia Telephony voice call the UE performs correct exchange of SIP protocol signalling messages; and

2) To verify that within SIP signalling the UE performs correct SDP offer/answer exchanges for negotiating media and indicating preconditions for resource reservation (as described by TS 24.229 [10], clause 6.1); and

3) To verify that when removing the video component from the IMS Multimedia Telephony call the UE performs correct exchange of SIP and SDP protocol messages.

H.17.2.4 Method of test

Initial conditions

UE is configured with the home domain name, public and private user identities and SIP Digest Credentials.

SS is configured with the home domain name, public and private user identities and SIP Digest Credentials. SS is listening to SIP default port 5060 for both UDP and TCP protocols. SS is able to perform MD5 authentication algorithm for that IMPI, according to TS 33.203 [14] clause 6.1 and RFC 3310 [17]. SS has performed MD5 authentication with the UE and accepted the registration. SS has also performed the UE Terminating voice Call according to C.11c.

Expected sequence

Step

Direction

Message

Comment

UE

SS

1

🡨

INVITE

SS sends re-INVITE with second SDP offer to add video.

2

🡪

100 Trying

(Optional) The UE responds with a 100 Trying provisional response.

3-5

Void

6

Make UE accept the speech and video offer.

(this is done either after receiption of 100 Trying or after 5s)

7

🡪

200 OK

The UE responds to the re-INVITE with a 200 OK final response.

8

🡨

ACK

The SS acknowledges the receipt of 200 OK for the re-INVITE.

9

🡨

INVITE

SS sends a re-INVITE with a SDP offer indicating that the video component is removed from the call.

10

🡪

100 Trying

(Optional) The UE responds with a 100 Trying provisional response.

11

🡪

200 OK

The UE responds to the re-INVITE with a 200 OK final response.

12

🡨

ACK

The SS acknowledges the receipt of 200 OK for INVITE.

13

🡨

BYE

The SS sends BYE to release the call.

14

🡪

200 OK

The UE sends 200 OK for the BYE request and ends the call.

NOTE: The default messages contents in annex A are used with condition "SIP Digest without TLS for Fixed Broadband Access" when applicable.

Specific Message Contents

INVITE (Step 1)

Use the default message "INVITE for MT Call" in clause A.2.9 with the following exceptions:

Header/param

Value/remark

Message-body

The following SDP types and values.

Session description:

  • v=0
  • o=- 1111111111 1111111111 IN (addrtype) (unicast-address for SS)
  • s=-
  • c=IN (addrtype) (connection-address for SS)
  • b=AS: 352

Time description:

  • t=0 0

Media description:

  • m=audio (transport port) RTP/AVP 99
  • b=AS:37
  • b=RS:0
  • b=RR:2000

Attributes for media:

  • a=rtpmap:99 AMR/8000/1
  • a=fmtp:99 mode-change-capability=2; max-red=220
  • a=ptime:20
  • a=maxptime:240

Media description:

– m=video (transport port) RTP/AVPF 101

– b=AS: 315

– b=RS: 0

– b=RR: 2500

Attributes for media:

– a=rtpmap:101 H264/90000

– a=fmtp: 101 packetization-mode=0;profile-level-id=42e00c; \

sprop-parameter-sets=J0LgDJWgUH6Af1A=,KM46gA==

– a=rtcp-fb:* trr-int 5000

– a=rtcp-fb:* nack

– a=rtcp-fb:* nack pli

– a=rtcp-fb:* ccm fir

– a=rtcp-fb:* ccm tmmbr

200 OK (Step 7)

Use the default message “200 OK for other requests than REGISTER or SUBSCRIBE” in annex A.3.1 with the following exceptions:

Header/param

Value/remark

Content-Type

media-type

application/sdp

Message-body

The following SDP types and values shall be present. [Note 3]

Session description:

  • v=0
  • o=(user-name) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE)
  • s=(session name)
  • c=IN (addrtype) (connection-address for UE) [Note 1]
  • b=AS: (bandwidth-value)

Time description:

  • t=0 0

Media description:

  • m=audio (transport port) RTP/AVP (fmt) [Note 2]
  • c=IN (addrtype) (connection-address for UE) [Note 1]
  • b=AS: (bandwidth-value)
  • b=RS: (bandwidth-value)
  • b=RR: (bandwidth-value)

Attributes for media:

  • a=rtpmap:(payload type) AMR/8000 [Note 2]
  • a=fmtp:(format) [Note 2]

Media description:

– m=video (transport port) RTP/AVPF (fmt) [Note 2]

– b=AS: (bandwidth-value)

– b=RS: (bandwidth-value)

– b=RR: (bandwidth-value)

Attributes for media:

– a=rtpmap: (payload type) H264/90000 [Note 2]

– a=fmtp: (format) packetization-mode=0;profile-level-id=(att-field); \

Note 1: At least one "c=" field shall be present.

Note 2: The value for fmt, payload type and format is not checked

Note 3: Parameters for the AMR codec are not checked

INVITE (Step 9)

Use the default message "INVITE for MT Call" in clause A.2.9 with the following exceptions:

Header/param

Value/remark

Message-body

The following SDP types and values.

Session description:

  • v=0
  • o=- 1111111111 1111111111 IN (addrtype) (unicast-address for SS)
  • s=-
  • c=IN (addrtype) (connection-address for SS)
  • b=AS:37

Time description:

  • t=0 0

Media description:

  • m=audio (transport port) RTP/AVP 99
  • b=AS:37
  • b=RS:0
  • b=RR:2000

Attributes for media:

  • a=rtpmap:99 AMR/8000/1
  • a=fmtp:99 mode-change-capability=2; max-red=220
  • a=ptime:20
  • a=maxptime:240

Media description:

– m=video (transport port) RTP/AVPF 101

– b=AS: 315

– b=RS: 0

– b=RR: 2500

Attributes for media:

– a=rtpmap: 101 H264/90000

– a=fmtp: 101packetization-mode=0;profile-level-id=42e00c; \

sprop-parameter-sets=J0LgDJWgUH6Af1A=,KM46gA==

– a=rtcp-fb:* trr-int 5000

– a=rtcp-fb:* nack

– a=rtcp-fb:* nack pli

– a=rtcp-fb:* ccm fir

– a=rtcp-fb:* ccm tmmbr

200 OK (Step 12)

Use the default message "200 OK for other requests than REGISTER or SUBSCRIBE" in clause A.3.1 with the following exceptions when there is no SDP in 180 Ringing.

Header/param

Value/remark

Require

option-tag

precondition

Content-Type

media-type

application/sdp

Content-Length

header shall be present if UE uses TCP to send this message and if there is a message body

value

length of message-body

Message-body

SDP body not checked.

H.17.2.5 Test requirements

The UE shall send requests and responses as described in clause H.17.2.4

Annex I (normative): IMS for Converged IP Communications