Re: RTP range issue on SIP Trunk CME

s.ranjbari · ‎03-23-2018

I have some issue with incoming calls and outgoing calls with sip trunk provider, all of the processes for initial session sip trunk can work but to choose RTP port it had issue (no voice, no ringback tone in my cell phone, and it show dialing, same time in router calls come), I checked it with asterisk family and it can work very good but for Cisco, it doesn't work good, I checked many commands like MTP and transcoder and NTE and min-sec etc,

in final with many trying I found it (with change ios when RTP port start with 16000 every call can work but for outgoing calls it changes RTP port to 18000 and can't work again, in another (most of it) version ios of cisco RTP port starts from 18000 and sometimes 19000, and issue again occurs

I am trying to find some command that with those can choose and specific RTP port range for CME, i found some command but it can't work on CME just works on CUBE

i found this command for change RTP range, BUT

Device(config)# voice service voip

Device(conf-voi-serv)# allow-connections sip to sip

Device(config-voi-serv)# media-address range 2001:DB8::/48

Device(config-voi-serv)# rtp-port range 20000 30000

But it doesn't work on CME ISR G1 like (2811), and in this issue,

is there anyone that can give any suggestion or command that can help??

Dennis Mink · ‎03-24-2018

Can you change the rtp.conf in asterisk to use 16k-32k ports for RTP.

those are pretty much default.

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s.ranjbari · ‎03-24-2018

I checked our sip trunk link with Asterisk, Asterisk isn't other switch, it was just for test link,
in fact, I have one CME Cisco (2811) in my border network, and one sip trunk link from sip provider like callcenteric, I have to change my setting to best setting, like change RTP port to 16000.
all of the outgoing and incoming calls come from this sip trunk link.

R0g22 · ‎03-24-2018

How did you isolate that the voice issue with the CME will be fixed by altering the port change ?
IOS will choose a random UDP port for RTP on which it will receive RTP and will convey it to the other end through SDP.
Next, those commands were introduced starting IOS 15.4(1)T or XE 3.9S as part of phantom packet monitoring feature. ISR-G1 won't have those commands.

s.ranjbari · ‎03-24-2018

just incoming can work but for outgoing it doesn't work , because it choose another port,

and it's results of it.

#Show sip-ua calls

15.X (can't work)

Call 1
SIP Call ID                : isbcihmxuuwusvcj33mvgcnxqt5uw5isijxu@SoftX3000
   State of the call       : STATE_SENT_ALERTING (14)
   Substate of the call    : SUBSTATE_NONE (0)
   Calling Number          : 9378378583
   Called Number           : 49085
   Bit Flags               : 0x8C4001C 0x10000140 0x444
   CC Call ID              : 78
   Source IP Address (Sig ): 10.194.22.200
   Destn SIP Req Addr:Port : [10.123.101.100]:5060
   Destn SIP Resp Addr:Port: [10.123.101.100]:5060
   Destination Name        : 10.123.101.100
   Number of Media Streams : 1
   Number of Active Streams: 0
   RTP Fork Object         : 0x0
   Media Mode              : flow-through
   Media Stream 1
     State of the stream      : STREAM_ADDING
     Stream Call ID           : 78
     Stream Type              : voice-only (0)
     Stream Media Addr Type   : 1
     Negotiated Codec         : g711alaw (160 bytes)
     Codec Payload Type       : 8
     Negotiated Dtmf-relay    : inband-voice
     Dtmf-relay Payload Type : 0
     QoS ID                   : -1
     Local QoS Strength       : BestEffort
     Negotiated QoS Strength : BestEffort
     Negotiated QoS Direction : None
     Local QoS Status         : None
     Media Source IP Addr:Port: [10.194.22.200]:18384
     Media Dest IP Addr:Port : [10.123.101.100]:20002

******************************************************************

12.4 (just incoming works)

Call 1
SIP Call ID                : isbc55xs3tig5nixi35hiwnv5mnjuvvij5xn@SoftX3000
   State of the call       : STATE_DEAD (10)
   Substate of the call    : SUBSTATE_NONE (0)
   Calling Number          : 9378378583
   Called Number           : 49085
   Bit Flags               : 0x884008C 0x300 0x800004
   CC Call ID              : 11
   Source IP Address (Sig ): 10.194.22.200
   Destn SIP Req Addr:Port : [10.123.101.100]:5060
   Destn SIP Resp Addr:Port: [10.123.101.100]:5060
   Destination Name        : 10.123.101.100
   Number of Media Streams : 1
   Number of Active Streams: 0
   RTP Fork Object         : 0x0
   Media Mode              : flow-through
   Media Stream 1
     State of the stream      : STREAM_DEAD
     Stream Call ID           : 11
     Stream Type              : voice-only (0)
     Stream Media Addr Type   : 1
     Negotiated Codec         : g711alaw (160 bytes)
     Codec Payload Type       : 8
     Negotiated Dtmf-relay    : inband-voice
     Dtmf-relay Payload Type : 0
     QoS ID                   : -1
     Local QoS Strength       : BestEffort
     Negotiated QoS Strength : BestEffort
     Negotiated QoS Direction : None
     Local QoS Status         : None
     Media Source IP Addr:Port: [10.194.22.200]:16858
     Media Dest IP Addr:Port : [10.123.101.100]:48458

and please check this picture in this picture after choosing the codec i think for choise RTP port it can't be success and occur error 500

R0g22 · ‎03-24-2018

Can you attach that pcap here please ? 500 error has nothing to do with RTP port.

s.ranjbari · ‎03-24-2018

Yes sure,

PFA

tnq

(just incoming works, outgoing can't work), with SBC softx3000 Huawei

R0g22 · ‎03-26-2018

Who is 10.123.101.165 and 10.194.22.222 ?

s.ranjbari · ‎03-27-2018

10.123.101.165 is SBC of Provider
10.194.22.222 our IP PBX