Real Time Protocol
RTP protocol is used to carry the voice and video streams between the endpoints. As mentioned in other sections, in CUCM system, RTP stream will flow directly between endpoints. However, in CME system, RTP stream will flow directly between endpoints ONLY if both endpoints are local to CME. In case one endpoint resides in different cluster, CME will act as MTP/CUBE between both endpoints (i.e, Endpoint A <-----> CME <-----> Endpoint B).
RTP protocol is UDP based with RTP header attached to it. The payload of RTP packet is the audio/video signals.
RTP header adds extra features over UDP protocol which are time stamp, sequence numbers, delivery monitoring. Those features allow the receiving endpoint to buffer RTP packets, reorder their sequence, remove the jitter, and play back as continuous stream.
Then if those features are required, why to use UDP and add extra RTP header instead of TCP??? Those features are inherited in TCP ???
Because TCP protocol uses flow control mechanism and retransmission. Each packet should be acknowledged. If not, then it will be retransmitted. This add extra delay to voice/video stream. In VoIP environments, packet loss can be accepted but not delay. This is provided by UDP protocol in addition to RTP.
The Payload Type (PT) value in RTP header indicate the codec used in the payload (to compress the audio/video stream). Codecs are covered in details in different section. Below is the mapping between PT values and codecs as per IANA.
PT encoding name audio/video (A/V) clock rate (Hz) channels (audio) Reference
-------- -------------- ----------------- --------------- ---------------- ---------
0 PCMU A 8000 1 [RFC3551]
3 GSM A 8000 1 [RFC3551]
4 G723 A 8000 1 [Kumar][RFC3551]
5 DVI4 A 8000 1 [RFC3551]
6 DVI4 A 16000 1 [RFC3551]
7 LPC A 8000 1 [RFC3551]
8 PCMA A 8000 1 [RFC3551]
9 G722 A 8000 1 [RFC3551]
10 L16 A 44100 2 [RFC3551]
11 L16 A 44100 1 [RFC3551]
12 QCELP A 8000 1 [RFC3551]
13 CN A 8000 1 [RFC3389]
14 MPA A 90000 [RFC3551][RFC2250]
15 G728 A 8000 1 [RFC3551]
16 DVI4 A 11025 1 [DiPol]
17 DVI4 A 22050 1 [DiPol]
18 G729 A 8000 1 [RFC3551]
19 Reserved A
20 Unassigned A
21 Unassigned A
22 Unassigned A
23 Unassigned A
24 Unassigned V
25 CelB V 90000 [RFC2029]
26 JPEG V 90000 [RFC2435]
27 Unassigned V
28 nv V 90000 [RFC3551]
29 Unassigned V
30 Unassigned V
31 H261 V 90000 [RFC4587]
32 MPV V 90000 [RFC2250]
33 MP2T AV 90000 [RFC2250]
34 H263 V 90000 [Zhu]
35-71 Unassigned ?
72-76 Reserved for RTCP conflict avoidance [RFC3551]
77-95 Unassigned ?
96-127 dynamic ? [RFC3551]
RTP protocol is using dynamic UDP port numbers ranging from 16384 to 32767. RTP protocol will use the even ports within the range while the odd ports are used for RTCP which is covered in next section.
Note: In streaming environment, RTP protocol is used in conjunction with RTSP protocol. RTP will carry the media while RTSP will control the streaming server (establish and control media sessions such as play and pause). This isn't available in IP Telephony environment.
In CUCM, you can control the port range in two places (this can't be done at GW level):
- SIP Profile
- Phone Configuration
Real Time Control Protocol
RTCP is engaged with RTP protocol. Each active RTP steam will have RTCP stream. RTCP stream will use the next odd UDP port (e.g. RTP is using UDP 18384 and RTCP will use 18385). Endpoints will exchange RTCP streams periodically. RTCP payload carries statistics about RTP reported from each endpoint. Below are the info provided by RTCP:
- Bytes sent
- Packets sent
- Lost packets
- Round trip delay
RTCP packet types:
- Sender Report Packet
- Receiver Report Packet
- Source Description RTCP Packet
- Good Bye RTCP packet
- Application-Specific RTCP Packets
At gateways (H323/SIP/MGCP) level, RTCP timer can be tuned using the command ip rtcp report interval. In CUCM RTCP can be disabled/enabled per device configuration (Device > Phone > #Select Phone# > Product Specific Configuration Layout > RTCP).
RTCP passthrough will be supported for SWMTP, TRP, RSVP and HWMTP in passthrough mode (codec passthrough). This is applicable to CUCM 9.0
- CME call-flows are not supported for RTCP passthrough.
- RTCP for transcoding call-flows is not supported.
- RTCP is not supported for conference calls.
To verify the statistics provided by RTCP,
- At Gateway level, use the command show call active voice brief or show voip rtp connections
- At Phones level, double click on ? Button or from the browser type the IP address of the phone
CSCtj37861 Bug Details
RTCP is sent by CIPC even though it is disabled in CUCM
CIPC Administration Guide suggests that RTCP is disabled by default but phone still sends out RTCP.
In CUCM version 6.x, device specific configuration for enabling or disabling RTCP for CIPC is not available
Test on CUCM version 7.x, call between CIPC 7.0.(3) and 7941 IP Phone with RTCP enabled on both shows RTCP.
- If RTCP on both phones are disabled from packet captures we can still see that RTCP is being sent by CIPC.
none identified so far
Hope this helps !! Remaining are Compressed RTP and Secured RTP which will be covered later in QoS & Voice Security posts.