[SIPForum-discussion] Sip audio negotiation problem between PBX and MCU (audioconferences)

Wed May 6 23:20:03 UTC 2015

Hello

Try to align the ptimes supported by the clients and the MCU for the same
codecs. Although the endpoints are not restricted to use different ptimes
for the payload exchange, it seems quite possible that this causes what you
say.

Also, about the DTMFs, you use RFC2833 (payload type 101 in the SDP), they
are not transmitted over g729. It is possible to do so but most likely the
tones would be of bad quality (thus unrecognized by the receiver) due to
compression.

About  the ptime, it is as you describe. When you do not see any ptime
declaration by the offerer, then the offerer use the by RFC default ptime =
20msec.

Best Regards
Dimitris

On Fri, May 1, 2015 at 10:34 PM, Antonio Manchon Romero <
antoniomanchonromero at yahoo.es> wrote:

> Hi Sumedha,
>
> I´m in contact with engineers on charge of those devices. And I´m going to
> explain this scenario.
> And I´d need to understand better how sip works.
> how would they change the priority of codecs configuration as preference,
> at sip INVITE SDP?
>
> Does sip message by itshelf show a codec preference?
> In Example:
> when invite sdp advices codecs like this:
>
> a=rtpmap:0 PCMU/8000
> a=ptime:30
> a=rtpmap:8 PCMA/8000
> a=ptime:30
> a=rtpmap:116 iLBC/8000
> a=ptime:20
>
> Is this message saying to remote endpoint that PCMU is the favorit, and
> iLBC the least in preference?
>
> In other hands, scenario sometimes needs DSP resources to make
> transcoding, and it´s needed to have end points sending packets with same
> packetization. So we´d need endpoints to advice 30 ms.
>
> How would they´ve to advice this packetization at SDP.
> I´ve seen SDP where it´s showd just the codec, but for others, it´s showed
> ptime or pmaxtime field.
>
> IE:
> a=rtpmap:0 PCMU/8000
> a=ptime:30
> a=rtpmap:8 PCMA/8000
> (missing ptime field) --> Does this mean PCMA adviced by endpoint could
> have any packetization time?, or it inherits last ptime showed for PCMU,
> 30ms.
> a=rtpmap:116 iLBC/8000
> a=ptime:20
>
> In other hands, for pmaxtime field, if it appears just once after a codec
> advice, this means this advice of pmaxtime is only for codec below
>
> In Example:
>
> a=rtpmap:0 PCMU/8000
> a=ptime:30
> a=rtpmap:8 PCMA/8000
> a=pmaxtime:40 ( does this field only apply to PCMA?, or remote endpoint
> will understand it could be used pmaxtime 40 ms only if PCMA is chosen?
> a=rtpmap:116 iLBC/8000
> a=ptime:20
>
> Thank you,
>
> regards,
>
> Antonio
>   ------------------------------
>  *De:* Sumedha Gawarikar <s.gawarikar at gmail.com>
> *Para:* Antonio Manchon Romero <antoniomanchonromero at yahoo.es>
> *CC:* discussion at sipforum.org
> *Enviado:* Jueves 30 de abril de 2015 20:15
> *Asunto:* Re: [SIPForum-discussion] Sip audio negotiation problem between
> PBX and MCU (audioconferences)
>
> It may be solved by changing the priority of codecs configuration as
> preference either in phones or system.
> Best Regards
> On Apr 30, 2015 9:41 PM, "Antonio Manchon Romero" <
> antoniomanchonromero at yahoo.es> wrote:
>
>
>
> Hi,
>
> I´d need help to troubleshoot a negotiation issue, were a phone calls to a
> MCU to make an audio connection.
> Call´s stablished at G729 30ms. And audio and dtmf tones are correct.
> But for some phones I´d prefer G722 or G711ulaw or G711alaw instead of
> G729.
>
> I´m integrating two systems for audio calls, a sip PBX and an Proxy Server
> vía sip trunk. Proxy Server is a registrar server for MCU
> (audioconferences).
>
> sip Phone supports: G722, PCMU, PCMA, G729 and other protocols.
> MCU supports: G7221, SIREN22, G729, G722, opus, PCMA, PCMU
>
> PBX has been configured to allow any audio codec with 64kpbs as maximum
> bandwith compression.  And PBX has forced packetization time to 30 ms.
> Sip-trunk from PBX to Proxy Server is configured with early offer as needed
> for calls from determined places to MCU.
>
> Call flow: phone --> PBX (sip-trunk Early Offer) --> Proxy Server --> MCU
>
> Here I paste sip traces taken at PBX. In this example call has connected
> with G729 30ms, and then it´s been disconnected normally.
>
> Some fields have been substituted by descripting names:
>
> Calling_Phone_Name
> Calling_Number
> called_number
> PBX_signalling_ip_address
> proxy_server_ip_address
> MCU_signalling_ip_address
> MCU_media_ip_address
>
> INVITE sip:called_number at proxy_server_ip_address:5060 SIP/2.0
> Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a6ba75c4b95b
> From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> To: <sip:called_number at proxy_server_ip_address>
> Date: Thu, 30 Apr 2015 10:10:57 GMT
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> Supported: timer,resource-priority,replaces
> Min-SE:  180
> User-Agent: PBX
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY
> CSeq: 101 INVITE
> Expires: 180
> Allow-Events: presence
> Supported: X-pbx-srtp-fallback,X-pbx-original-called
> Call-Info:
> <sip:PBX_signalling_ip_address:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
> pbx-Guid: 0808637440-0000065536-0000144815-0202899735
> Session-Expires:  1800
> P-Asserted-Identity: "Calling_Phone_Name"
> <sip:Calling_Number at PBX_signalling_ip_address>
> Remote-Party-ID: "Calling_Phone_Name"
> <sip:Calling_Number at PBX_signalling_ip_address
> >;party=calling;screen=yes;privacy=off
> Contact: <sip:Calling_Number at PBX_signalling_ip_address:5060>
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 406
>
> v=0
> o=pbxSystemsCall_Agent-SIP 865212 1 IN IP4 PBX_signalling_ip_address
> s=SIP Call
> c=IN IP4 Calling_Phone_ip_address
> b=TIAS:64000
> b=AS:64
> t=0 0
> m=audio 20488 RTP/AVP 0 8 116 18 101
> a=rtpmap:0 PCMU/8000
> a=ptime:30
> a=rtpmap:8 PCMA/8000
> a=ptime:30
> a=rtpmap:116 iLBC/8000
> a=ptime:20
> a=maxptime:100
> a=fmtp:116 mode=20
> a=rtpmap:18 G729/8000
> a=ptime:30
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
>
> SIP/2.0 180 Ringing
> CSeq: 101 INVITE
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
> Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a6ba75c4b95b
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
> CSeq: 101 INVITE
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
> Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a6ba75c4b95b
> Allow-Events: conference,refer,conference
> User-Agent: MCU_version
> P-RMX-Info: m,c,128000,72,a
> Allow:
> INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,UPDATE,PRACK,SUBSCRIBE,NOTIFY,BENOTIFY
> Require: timer
> Supported:
> X-pbx-callinfo,plcm-ivr-service-provider,ms-early-media,replaces,resource-priority,histinfo,ms-safe-transfer,X-pbx-sis-5.1.0,tdialog,timer,100rel,X-pbx-srtp-fallback,ms-conf-invite,ms-sender
> Contact: "inbound" <sip:called_number at proxy_server_ip_address
> :5060;transport=udp>
> Content-Type: application/sdp
> Session-Expires: 1800;refresher=uac
> Content-Length: 262
>
> v=0
> o=- 1430388657 88866816 IN IP4 MCU_signalling_ip_address
> s=SIP Call
> c=IN IP4 MCU_media_sip_address
> b=AS:8
> t=0 0
> m=audio 49288 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=maxptime:30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=vnd.polycom.PlcmMaskCap:0011
> a=sendrecv
>
>
> ACK sip:called_number at proxy_server_ip_address:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a6bd7cc5bf3c
> From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
> Date: Thu, 30 Apr 2015 10:10:57 GMT
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> Max-Forwards: 70
> CSeq: 101 ACK
> Allow-Events: presence
> Content-Length: 0
>
>
>
> INVITE sip:Calling_Number at PBX_signalling_ip_address:5060 SIP/2.0
> Via: SIP/2.0/UDP
> proxy_server_ip_address:5060;branch=z9hG4bK-3739-3ca9e7aba03c8134034c251595d8e028
> CSeq: 1 INVITE
> From: <sip:called_number at proxy_server_ip_address>;tag=634f26de
> To: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> Allow-Events: conference,refer,conference
> User-Agent: MCU_version
> P-RMX-Info: i,c,128000,32,a
> Allow:
> INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,UPDATE,PRACK,SUBSCRIBE,NOTIFY,BENOTIFY
> Supported:
> X-pbx-callinfo,ms-early-media,plcm-ivr-service-provider,replaces,resource-priority,histinfo,ms-safe-transfer,X-pbx-sis-5.1.0,tdialog,timer,100rel,X-pbx-srtp-fallback,ms-conf-invite,ms-sender,tdialog
> Referred-By: <sip:called_number at proxy_server_ip_address
> :5060;/vmr;transport=UDP>
> Plcm-Call-ID: 6ac0899a-0b0d-412f-852e-5e3f00db515b
> Contact: "inbound" <sip:called_number at proxy_server_ip_address
> :5060;transport=udp>;isfocus
> Max-Forwards: 69
> Content-Type: application/sdp
> Session-Expires: 1800;refresher=uas
> Min-Expires: 90
> Content-Length: 875
>
> v=0
> o=- 1430388657 88866817 IN IP4 MCU_signalling_ip_address
> s=rmx2k Conf
> c=IN IP4 MCU_media_sip_address
> b=AS:32
> t=0 0
> m=audio 49290 RTP/AVP 104 105 114 113 103 102 18 9 127 8 0 98 101
> a=rtpmap:104 G7221/16000
> a=fmtp:104 bitrate=32000
> a=rtpmap:105 SIREN22/48000
> a=fmtp:105 bitrate=32000
> a=rtpmap:114 G7221/32000
> a=fmtp:114 bitrate=32000
> a=rtpmap:113 G7221/32000
> a=fmtp:113 bitrate=24000
> a=rtpmap:103 G7221/16000
> a=fmtp:103 bitrate=24000
> a=rtpmap:102 G7221/16000
> a=fmtp:102 bitrate=16000
> a=rtpmap:18 G729/8000
> a=maxptime:30
> a=rtpmap:9 G722/8000
> a=rtpmap:127 opus/48000/2
> a=fmtp:127 sprop-maxplaybackrate=48000; maxaveragebitrate=32000;
> sprop-stereo=0; stereo=0
> a=ptime:1
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=rtpmap:0 PCMU/8000
> a=rtpmap:98 SIREN14/16000
> a=fmtp:98 bitrate=32000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=vnd.polycom.PlcmMaskCap:0011
> a=sendrecv
>
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> proxy_server_ip_address:5060;branch=z9hG4bK-3739-3ca9e7aba03c8134034c251595d8e028
> From: <sip:called_number at proxy_server_ip_address>;tag=634f26de
> To: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> Date: Thu, 30 Apr 2015 10:11:09 GMT
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> CSeq: 1 INVITE
> Allow-Events: presence
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> proxy_server_ip_address:5060;branch=z9hG4bK-3739-3ca9e7aba03c8134034c251595d8e028
> From: <sip:called_number at proxy_server_ip_address>;tag=634f26de
> To: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> Date: Thu, 30 Apr 2015 10:11:09 GMT
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> CSeq: 1 INVITE
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY
> Allow-Events: presence
> Supported: replaces
> Supported: X-pbx-srtp-fallback
> Supported: Geolocation
> Session-Expires:  1800;refresher=uas
> Require:  timer
> P-Asserted-Identity: "Calling_Phone_Name"
> <sip:Calling_Number at PBX_signalling_ip_address>
> Remote-Party-ID: "Calling_Phone_Name"
> <sip:Calling_Number at PBX_signalling_ip_address
> >;party=called;screen=yes;privacy=off
> Contact: <sip:Calling_Number at PBX_signalling_ip_address:5060>
> Content-Type: application/sdp
> Content-Length: 243
>
> v=0
> o=pbxSystemsCall_Agent-SIP 865212 2 IN IP4 PBX_signalling_ip_address
> s=SIP Call
> c=IN IP4 Calling_Phone_ip_address
> b=AS:8
> t=0 0
> m=audio 20488 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=ptime:30
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
>
> ACK sip:Calling_Number at PBX_signalling_ip_address:5060 SIP/2.0
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> CSeq: 1 ACK
> Via: SIP/2.0/UDP
> proxy_server_ip_address:5060;branch=z9hG4bK-3739-a27c389a0eaf38ae5e8d6be5c2d1f79c
> From: <sip:called_number at proxy_server_ip_address>;tag=634f26de
> To: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> Max-Forwards: 70
> Content-Length: 0
>
>
>
> BYE sip:called_number at proxy_server_ip_address:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a88b1d3ee38
> From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
> Date: Thu, 30 Apr 2015 10:11:09 GMT
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> User-Agent: PBX
> Max-Forwards: 70
> P-Asserted-Identity: "Calling_Phone_Name"
> <sip:Calling_Number at PBX_signalling_ip_address>
> CSeq: 102 BYE
> Reason: Q.850;cause=16
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
> CSeq: 102 BYE
> Call-ID: 3032d400-5411ffb1-273ab-c180117 at PBX_signalling_ip_address
> From: "Calling_Phone_Name" <sip:Calling_Number at PBX_signalling_ip_address
> >;tag=865212~e390a5a7-d5c3-4e13-afd1-f8f73d1be093-50102620
> To: <sip:called_number at proxy_server_ip_address>;tag=634f26de
> Via: SIP/2.0/UDP PBX_signalling_ip_address:5060;branch=z9hG4bK6a88b1d3ee38
> Content-Length: 0
>
> ***********************************************
>
> Thanks for your help in advance,
>
> Regards,
>
>
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20150507/50a9cb0c/attachment-0002.html>