[Asterisk-Users] Cisco to Cisco - poor quality

[TeleSIP]

Hi Terence,

I can take a look at the traces if you want.  Just repeat the test using
g711ulaw and use Ethereal to capture the SIP messages and RTP stream of the
phone that hears bad sound, and if you can, of the other phone too (the one
that hears fine).  Send me the captures and I will see if there is some
obvious problem.

Regards,
Andres
http://www.telesip.net

----- Original Message ----- 
From: "Terence Parker" <terence at parker.com.hk>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, January 07, 2004 8:38 PM
Subject: Re: [Asterisk-Users] Cisco to Cisco - poor quality


> I have managed to find time to have another go at the Cisco phones -
> alas, I am still having problems with Cisco to Cisco calls.
>
> Just to re-cap (it's been a few days!) i'm using Cisco 7960's and have
> tried setting both phones to different codecs (tried default g729a,
> g711alaw, and g711ulaw). Also, the other observations that have been
> made:
>
> - Problem is one-way. One side hears me clearly ; I don't hear the
> other side clearly at all (5% audible only).
> - Calls to MSN are fine (two way conversation is crystal clear)
> - Calls to a Zultys Zip2 SIP phone is also perfectly clear.
> - All these three tested over the same network and same VPN (call
> between Hong Kong and USA).
> - Cisco to Cisco calls worked fine with Vocal.
>
> If Cisco is able to talk fine with other devices, there should not be a
> problem with bandwidth or my network. However, I am finding it quite
> bizzarre that Cisco is unable to talk to itself. The problem shouldn't
> be VAD or the like - even if I talk non-stop, or the other guy does, I
> get the same problem.
>
> I attach a copy of my Cisco phone configuration for reference. I have
> even recently upgraded my phone firmware - but no luck.
>
> Platform : Cisco IP Phone 7960
> Elasped Time: 08:11:26
>
> dhcp_server : 192.168.8.254
> my_ip_addr : 192.168.8.83
> subnet_mask : 255.255.255.0
> defaultgw : 192.168.8.254
> dyn_dns_addr_1 : 0.0.0.0
> dyn_dns_addr_2 : 0.0.0.0
> dns_addr : 205.252.144.228
> dns_backup_1: 202.14.67.4
> tftp_addr : 192.168.0.252
> dyn_tftp_addr : 0.0.0.0
> my_mac_addr : 0007:50ac:6932
> domain_name : deltapath.com
> my_name : SIP000750AC6932
> Status Flags : 12300000
>
> image_version : "P0S3-05-3-00"
> FirmLoadID : "PC03A300"
> network_media_type : Auto
> network_port2_type : Hub/Switch
> tos_media : 5
> phone_label : "DELTAPATH"
> tftp_cfg_dir : "./sip_phone/"
> phone_password : **********
> phone_prompt : "SIP Phone"
> language : english
> sntp_mode : DirectedBroadcast
> sntp_server : stdtime.gov.hk
> time_zone : HST
> dst_offset : 0
> dst_start_month : April
> dst_start_day : 0
> dst_start_day_of_week : Sun
> dst_start_week_of_month : 1
> dst_start_time : 02
> dst_stop_month : Oct
> dst_stop_day : 0
> dst_stop_day_of_week : Sunday
> dst_stop_week_of_month : 8
> dst_stop_time : 2
> dst_auto_adjust : 0
> time_format_24hr : 1
> date_format : M/D/Y
> nat_enable : 0
> nat_address :
> voip_control_port : 5060
> start_media_port : 16384
> end_media_port : 32766
> sync : "1"
> xml_card_dir : ""
> xml_card_file : "CARD.XML"
> telnet_level : 2
> services_url : ""
> directory_url : ""
> logo_url : "http://deltapath.com/logo.bmp"
> http_proxy_addr :
> http_proxy_port : 80
> enable_vad : 0
> dial_template : "dialplan"
> callerid_blocking : 0
> anonymous_call_block : 0
> autocomplete : 1
> messages_uri : "86"
> dnd_control : 0
> preferred_codec : g729a
> dtmf_outofband : avt
> dtmf_avt_payload : 101
> dtmf_db_level : 3
> dtmf_inband : 1
> line1_name : "TerenceParker"
> line2_name : "74xxx"
> line3_name : "74xxx"
> line4_name : ""
> line5_name : ""
> line6_name : ""
> line1_authname : "TerenceParker"
> line2_authname : "74xxx"
> line3_authname : "74xxx"
> line4_authname : "UNPROVISIONED"
> line5_authname : "UNPROVISIONED"
> line6_authname : "UNPROVISIONED"
> line1_shortname : "Asterisk"
> line2_shortname : "FWD-74xxx"
> line3_shortname : "FWD-74xxx"
> line4_shortname : "UNPROVISIONED"
> line5_shortname : "UNPROVISIONED"
> line6_shortname : "UNPROVISIONED"
> line1_displayname : "TerenceParker"
> line2_displayname : "74xxx"
> line3_displayname : "Terence Parker"
> line4_displayname : ""
> line5_displayname : ""
> line6_displayname : ""
> proxy1_address : "192.168.0.254"
> proxy2_address : "fwd.pulver.com"
> proxy3_address : "fwd.pulver.com"
> proxy4_address : ""
> proxy5_address : ""
> proxy6_address : ""
> proxy1_port : 5060
> proxy2_port : 5060
> ........
> sip_retx : 10
> sip_invite_retx : 6
> timer_t1 : 500
> timer_t2 : 4000
> timer_invite_expires : 180
> timer_register_expires : 3600
> proxy_register : 1
> proxy_backup : "UNPROVISIONED"
> proxy_emergency : "UNPROVISIONED"
> proxy_backup_port : 0
> proxy_emergency_port : 0
> outbound_proxy :
> outbound_proxy_port : 5082
> nat_received_processing : 0
> mwi_status : 0
> call_waiting : 1
> user_info : none
> cnf_join_enable : 1
> remote_party_id : 0
> semi_attended_transfer : 1
> call_hold_ringback : 0
>
>
> Thanks for any help!
>
> Terence
>
>
> > I have never used Cisco phones, but I have had problems in the past
> > relating to * RTP talking to a widget with VAD turned on.
> > * RTP stack can not run on its own.  It relies on receiving RTP packets
> > for doing its timing.
> >
> > A simple test is to sniff the line to make sure the phones always send
> > packets.
> > If you see pauses, you may need to disable some type of VAD setting on
> > the phone.
> > Or just never quit talking when using the Cisco phone.
> >
> > Terence Parker wrote:
> >
> >> I have set canreinvite=no in the sip.conf for each user (well, there
> >> are
> >> only two) using a cisco phone. What does this imply?
> >>
> >> As for whether the problem is due to the phones or asterisk however,
> >> indications would suggest both, because:
> >>
> >> - Voicemail works fine (and is clear)
> >> - I can initiate a call between MSN and Cisco, and that would sound
> >> fine.
> >>
> >> This might suggest a problem with my phones. However :
> >>
> >>    -  When using Vocal previously, Cisco to Cisco conversation was
> >> fine.
> >>
> >> This has led me to be completely stumped! I notice some mention
> >> elsewhere
> >> about asterisk lacking certain codecs because of license
> >> restrictions? Is
> >> this anything to do with me? Or should the phones still - in theory -
> >> be
> >> able to talk to each other without any problems? I have tried the
> >> cisco
> >> phone on both g729a and g711ulaw.
> >>
> >> I'm currently *trying* to get ahold of an updated firmware for my
> >> phone. I
> >> will see if this fixes the problems.
> >>
> >> Thanks again,
> >>
> >> Terence
>
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